Csound Manual

The Public Csound Reference Manual
CANONICAL VERSION 4.10

by Barry Vercoe, Media Lab MIT & contributors
Edited by John ffitch, Richard Boulanger, Jean Pich, & David Boothe
Copyright 1986, 1992 by the Massachusetts Institute of Technology. All rights reserved.
Copyright Notice
Copyright 1986, 1992 by the Massachusetts Institute of Technology. All rights reserved. Developed by Barry L. Vercoe at the Experimental Music Studio, Media Laboratory, MIT, Cambridge, Massachusetts, with partial support from the System Development Foundation and from National Science Foundation Grant # IRI-8704665. Permission to use, copy, or modify these programs and their documentation for educational and research purposes only and without fee is hereby granted, provided that this copyright and permission notice appear on all copies and supporting documentation. For any other uses of this software, in original or modified form, including but not limited to distribution in whole or in part, specific prior permission from MIT must be obtained. MIT makes no representations about the suitability of this software for any purpose. It is provided as is without express or implied warranty The original Hypertext Edition of the MIT Csound Manual was prepared for the World Wide Web by Peter J. Nix of the Department of Music at the University of Leeds and Jean Pich of the Facult de musique de lUniversit de Montral. This Print Edition, in Adobe Acrobat format, and the current HTML Edition, are maintained by David M. Boothe of Lakewood Sound. The editors fully acknowledge the rights of the authors of the original documentation and programs, as set out above, and further request that this notice appear wherever this material is held.
The Public Csound Reference Manual Version 4.10
Preface
Page ii
Contributors
In addition to the core code developed by Barry Vercoe at MIT, a large part of the Csound code was modified, developed and extended by an independent group of programmers, composers and scientists. Copyright to this code is held by the respective authors:
Mike Berry Eli Breder Michael Casey Michael Clark Perry Cook Sean Costello Richard Dobson Mark Dolson Rasmus Ekman Dan Ellis Tom Erbe John ffitch Bill Gardner Matt Ingalls
Richard Karpen Victor Lazzarini Allan Lee David Macintyre Gabriel Maldonado Max Mathews Hans Mikelson Peter Neubcker Ville Pulkki Marc Resibois Rob Shaw Paris Smaragdis Greg Sullivan Bill Verplank Robin Whittle
This manual was compiled from the canonical Csound Manual sources maintained by John ffitch, Richard Boulanger, Jean Pich and David Boothe. The Acrobat Edition of this manual was redesigned for the Csound version 4.10 release, in February & March, 2001. It is set in 10 pt. Trebuchet, from Microsoft Corporation. Headings are set Antique Olive from Hewlett-Packard Corporation. Syntax and code examples are set in Andale Mono from Monotype Corporation.
Preface
Page iii
This page intentionally left blank.
Preface
Page iv
Table of Contents
COPYRIGHT NOTICE CONTRIBUTORS TABLE OF CONTENTS FINDER 1 PREFACE
1.1 1.2 1.3 1.4 Where to Get Public Csound and the Csound Manual How to Install Csound How to use the Csound Manual The Csound Mailing List
II
III
XV
11
13 14 18 19
2 SYNTAX OF THE ORCHESTRA

2.1 2.2 2.3 2.4 2.5 Directories and Files Nomenclature Orchestra Statement Types Constants and Variables Expressions
2-1
2-2 2-3 2-6 2-7 2-8
3 ORCHESTRA SYNTAX: ORCHESTRA HEADER STATEMENTS

3.1 3.2 3.3 3.4 3.5 sr, kr, ksmps, nchnls strset, pset seed ftgen massign, ctrlinit
3-1
3-1 3-2 3-3 3-4 3-5
4 ORCHESTRA SYNTAX: INSTRUMENT BLOCK STATEMENTS

4.1 instr, endin
4-1
4-1
5 ORCHESTRA SYNTAX: VARIABLE INITIALIZATION

5.1 =, init, tival, divz
5-1
5-1
6 INSTRUMENT CONTROL: INSTRUMENT INVOCATION

6.1 6.2 6.3 schedule, schedwhen schedkwhen turnon
6-1
6-1 6-3 6-4
Preface
Page v
7 INSTRUMENT CONTROL: DURATION CONTROL STATEMENTS

7.1 ihold, turnoff
7-1
7-1
8 INSTRUMENT CONTROL: REAL-TIME PERFORMANCE CONTROL

8.1 8.2 active cpuprc, maxalloc, prealloc
8-1
8-1 8-2
9 INSTRUMENT CONTROL: TIME READING

9.1 timek, times, timeinstk, timeinsts
9-1
9-1
10 INSTRUMENT CONTROL: CLOCK CONTROL

10.1 clockon, clockoff, readclock
10-1
10-1
11 INSTRUMENT CONTROL: SENSING AND CONTROL

11.1 11.2 11.3 11.4 11.5 11.6 11.7 11.8 11.9 11.10 11.11 pitch pitchamdf tempest follow trigger peak xyin, tempo follow2 setctrl, control button, checkbox sensekey
11-1
11-1 11-3 11-5 11-7 11-8 11-9 11-10 11-11 11-12 11-13 11-14
12 INSTRUMENT CONTROL: CONDITIONAL VALUES

12.1 >, <, >=, <=, ==, !=, ?
12-1
12-1
13 INSTRUMENT CONTROL: MACROS

13.1 13.2 #define, $NAME, #undef #include
13-1
13-1 13-3
14 INSTRUMENT CONTROL: PROGRAM FLOW CONTROL

14.1 igoto, tigoto, kgoto, goto, if, timout
14-1
14-1
15 INSTRUMENT CONTROL: REINITIALIZATION

15.1 reinit, rigoto, rireturn
15-1
15-1
16 MATHEMATICAL OPERATIONS: ARITHMETIC AND LOGIC OPERATIONS

16.1 -, +, &&, ||, *, /, ^, %
16-1
16-1
17 MATHEMATICAL OPERATIONS: MATHEMATICAL FUNCTIONS

17.1 17.2 int, frac, i, abs, exp, log, log10, sqrt powoftwo, logbtwo
Preface Page vi
17-1
17-1 17-2
18 MATHEMATICAL OPERATIONS: TRIGONOMETRIC FUNCTIONS

18.1 sin, cos, tan, sininv, cosinv, taninv, sinh, cosh, tanh
18-1
18-1
19 MATHEMATICAL OPERATIONS: AMPLITUDE FUNCTIONS

19.1 dbamp, ampdb dbfsamp, ampdbfs
19-1
19-1
20 MATHEMATICAL OPERATIONS: RANDOM FUNCTIONS

20.1 rnd, birnd
20-1
20-1
21 MATHEMATICAL FUNCTIONS: OPCODE EQUIVALENTS OF FUNCTIONS

21.1 21.2 21.3 21.4 21.5 sum product pow taninv2 mac, maca
21-1
21-1 21-2 21-3 21-4 21-5
22 PITCH CONVERTERS: FUNCTIONS

22.1 octpch, pchoct, cpspch, octcps, cpsoct
22-1
22-1
23 PITCH CONVERTERS: TUNING OPCODES

23.1 cps2pch, cpsxpch
23-1
23-1
24 MIDI SUPPORT: CONVERTERS

24.1 24.2
24-1
notnum, veloc, cpsmidi, cpsmidib, octmidi, octmidib, pchmidi, pchmidib, ampmidi, aftouch, pchbend, midictrl 24-1 cpstmid 24-3
25 MIDI SUPPORT: CONTROLLER INPUT

25.1 25.2 25.3 initc7, initc14, initc21 midic7, midic14, midic21, ctrl7, ctrl14, ctrl21 chanctrl
25-1
25-1 25-2 25-4
26 MIDI SUPPORT: SLIDER BANKS

26.1
26-1
slider8, slider16, slider32, slider64, slider8f, slider16f, slider32f, slider64f, s16b14, s32b14 26-1
27 MIDI SUPPORT: GENERIC I/O

27.1 27.2 midiin midiout
27-1
27-1 27-2
28 MIDI SUPPORT: NOTE-ON/NOTE-OFF

28.1 28.2 28.3 noteon, noteoff, noteondur, noteondur2 moscil, midion midion2
28-1
28-1 28-3 28-4
Preface
Page vii
29 MIDI SUPPORT: MIDI MESSAGE OUTPUT

29.1 29.2 29.3
29-1
outic, outkc, outic14, outkc14, outipb, outkpb, outiat, outkat, outipc, outkpc, outipat, outkpat 29-1 nrpn 29-3 mdelay 29-4
30 MIDI SUPPORT: REAL-TIME MESSAGES

30.1 mclock, mrtmsg
30-1
30-1
31 MIDI SUPPORT: EVENT EXTENDERS

31.1 xtratim, release
31-1
31-1
32 SIGNAL GENERATORS: LINEAR AND EXPONENTIAL GENERATORS

32.1 32.2 32.3 line, expon, linseg, linsegr, expseg, expsegr, expsega adsr, madsr, xadsr, mxadsr transeg
32-1
32-1 32-3 32-4
33 SIGNAL GENERATORS: TABLE ACCESS

33.1 table, tablei, table3, oscil1, oscil1i, osciln
33-1
33-1
34 SIGNAL GENERATORS: PHASORS

34.1 34.2 phasor phasorbnk
34-1
34-1 34-2
35 SIGNAL GENERATORS: BASIC OSCILLATORS

35.1 35.2 35.3 oscil, oscili, oscil3 poscil, poscil3 lfo
35-1
35-1 35-2 35-3
36 SIGNAL GENERATORS: DYNAMIC SPECTRUM OSCILLATORS

36.1 36.2 36.3 buzz, gbuzz vco mpulse
36-1
36-1 36-3 36-5
37 SIGNAL GENERATORS: ADDITIVE SYNTHESIS/RESYNTHESIS

37.1 37.2 37.3 adsyn adsynt hsboscil
37-1
37-1 37-3 37-5
38 SIGNAL GENERATORS: FM SYNTHESIS

38.1 38.2 38.3 foscil, foscili fmvoice fmbell, fmrhode, fmwurlie, fmmetal, fmb3, fmpercfl
38-1
38-1 38-2 38-3
Preface
Page viii
39 SIGNAL GENERATORS: SAMPLE PLAYBACK

39.1 39.2 39.3
39-1
loscil, loscil3 39-1 lposcil, lposcil3 39-3 sfload, sfplist, sfilist, sfpassign, sfpreset, sfplay, sfplaym, sfinstr, sfinstrm 39-4
40 SIGNAL GENERATORS: GRANULAR SYNTHESIS

40.1 40.2 40.3 40.4 40.5 fof, fof2 fog grain granule sndwarp, sndwarpst
40-1
40-1 40-3 40-5 40-7 40-10
41 SIGNAL GENERATORS: SCANNED SYNTHESIS

41.1 41.2 scanu scans
41-1
41-3 41-5
42 SIGNAL GENERATORS: WAVEGUIDE PHYSICAL MODELING

42.1 42.2 42.3 42.4 42.5 42.6 42.7 42.8 pluck wgpluck repluck, wgpluck2 wgbow wgflute wgbrass wgclar wgbowedbar
42-1
42-1 42-3 42-4 42-5 42-6 42-7 42-8 42-9
43 SIGNAL GENERATORS: MODELS AND EMULATIONS

43.1 43.2 43.3 43.4 43.5 43.6 43.7 43.8 43.9 43.10 moog shaker marimba, vibes mandol gogobel voice lorenz planet cabasa, crunch, sekere, sandpaper, stix guiro, tambourine, bamboo, dripwater, sleighbells
43-1
43-1 43-2 43-3 43-5 43-6 43-7 43-8 43-10 43-12 43-14
44 SIGNAL GENERATORS: STFT RESYNTHESIS (VOCODING)

44.1 44.2 44.3 pvoc, vpvoc pvread, pvbufread, pvinterp, pvcross, tableseg, tablexseg pvadd
44-1
44-1 44-3 44-6
45 SIGNAL GENERATORS: LPC RESYNTHESIS

45.1 45.2 lpread, lpreson, lpfreson lpslot, lpinterp
45-1
45-1 45-3
Preface
Page ix
46 SIGNAL GENERATORS: RANDOM (NOISE) GENERATORS

46.1 46.2 46.3 46.4 rand, randh, randi x-class noise generators pinkish noise
46-1
46-1 46-2 46-4 46-6
47 FUNCTION TABLE CONTROL: TABLE QUERIES

47.1 47.2 ftlen, ftlptim, ftsr, nsamp tableng
47-1
47-1 47-2
48 FUNCTION TABLE CONTROL: TABLE SELECTION

48.1 tablekt, tableikt
48-1
48-1
49 FUNCTION TABLE CONTROL: READ/WRITE OPERATIONS

49.1 49.2 49.3 tableiw, tablew, tablewkt tablegpw, tablemix, tablecopy, tableigpw, tableimix, tableicopy tablera, tablewa
49-1
49-1 49-3 49-5
50 SIGNAL MODIFIERS: STANDARD FILTERS

50.1 50.2 50.3 50.4 50.5 50.6 50.7 50.8 50.9 50.10 50.11 50.12 50.13 50.14 port, portk, tone, tonek, atone, atonek, reson, resonk, areson, aresonk tonex, atonex, resonx resonr, resonz resony lowres, lowresx vlowres lowpass2 biquad, rezzy, moogvcf svfilter hilbert butterhp, butterlp, butterbp, butterbr filter2, zfilter2 lpf18 tbvcf
50-1
50-1 50-3 50-4 50-7 50-8 50-9 50-10 50-12 50-14 50-16 50-19 50-20 50-22 50-23
51 SIGNAL MODIFIERS: SPECIALIZED FILTERS

51.1 51.2 51.3 nlfilt pareq dcblock
51-1
51-1 51-3 51-5
52 SIGNAL MODIFIERS: ENVELOPE MODIFIERS

52.1 linen, linenr, envlpx, envlpxr
52-1
52-1
53 SIGNAL MODIFIERS: AMPLITUDE MODIFIERS

53.1 53.2 53.3 rms, gain, balance dam clip
53-1
53-1 53-2 53-3
Preface
Page x
54 SIGNAL MODIFIERS: SIGNAL LIMITERS

54.1 limit, mirror, wrap
54-1
54-1
55 SIGNAL MODIFIERS: DELAY

55.1 55.2 55.3 55.4 delayr, delayw, delay, delay1 deltap, deltapi, deltapn, deltap3 multitap vdelay, vdelay3
55-1
55-1 55-3 55-5 55-6
56 SIGNAL MODIFIERS: REVERBERATION

56.1 56.2 56.3 56.4 comb, alpass, reverb reverb2, nreverb nestedap babo
56-1
56-1 56-3 56-5 56-7
57 SIGNAL MODIFIERS: WAVEGUIDES

57.1 57.2 wguide1, wguide2 streson
57-1
57-1 57-3
58 SIGNAL MODIFIERS: SPECIAL EFFECTS

58.1 58.2 58.3 58.4 harmon flanger distort1 phaser1, phaser2
58-1
58-1 58-3 58-4 58-6
59 SIGNAL MODIFIERS: CONVOLUTION AND MORPHING

59.1 59.2 convolve cross2
59-1
59-1 59-4
60 SIGNAL MODIFIERS: PANNING AND SPATIALIZATION

60.1 60.2 60.3 60.4 60.5
60-1
pan 60-1 locsig, locsend 60-3 space, spsend, spdist 60-5 hrtfer 60-9 vbaplsinit, vbap4, vbap8, vbap16, vbap4move, vbap8move, vbap16move, vbapz, vbapzmove 60-10
61 SIGNAL MODIFIERS: SAMPLE LEVEL OPERATORS

61.1 61.2 61.3 samphold, downsamp, upsamp, interp, integ, diff ntrpol fold
61-1
61-1 61-3 61-4
62 ZAK PATCH SYSTEM

62.1 62.2 62.3 62.4 zakinit ziw, zkw, zaw, ziwm, zkwm, zawm zir, zkr, zar, zarg zkmod, zamod, zkcl, zacl
62-1
62-2 62-3 62-5 62-6
Preface
Page xi
63 OPERATIONS USING SPECTRAL DATA TYPES

63.1 63.2 63.3 63.4 specaddm, specdiff, specscal, spechist, specfilt specptrk specsum, specdisp spectrum
63-1
63-2 63-3 63-5 63-6
64 SIGNAL INPUT AND OUTPUT: INPUT

64.1 64.2 in, ins, inq, inh, ino, soundin, diskin inx, in32, inch, inz
64-1
64-1 64-3
65 SIGNAL INPUT AND OUTPUT: OUTPUT

65.1 65.2
65-1
soundout, soundouts, out, outs1, outs2, outs, outq1, outq2, outq3, outq4, outq, outh, outo 65-1 outx, out32, outc, outch, outz 65-3
66 SIGNAL INPUT AND OUTPUT: FILE I/O

66.1 66.2 66.3 66.4 dumpk, dumpk2, dumpk3, dumpk4, readk, readk2, readk3, readk4 fout, foutk, fouti, foutir, fiopen fin, fink, fini vincr, clear
66-1
66-1 66-3 66-5 66-6
67 SIGNAL INPUT AND OUTPUT: SOUND FILE QUERIES

67.1 filelen, filesr, filenchnls, filepeak
67-1
67-1
68 SIGNAL INPUT AND OUTPUT: PRINTING AND DISPLAY

68.1 68.2 68.3 print, display, dispfft printk, printks printk2
68-1
68-1 68-2 68-4
69 THE STANDARD NUMERIC SCORE

69.1 69.2 69.3 69.4 69.5 69.6 69.7 69.8 69.9 69.10 69.11 69.12 69.13 69.14 69.15 69.16 69.17 Preprocessing of Standard Scores Next-P and Previous-P Symbols Ramping Score Macros Multiple File Score Evaluation of Expressions f Statement (or Function Table Statement) i Statement (Instrument or Note Statement) a Statement (or Advance Statement) t Statement (Tempo Statement) b Statement v Statement s Statement e Statement r Statement (Repeat Statement) m Statement (Mark Statement) n Statement
69-1
69-1 69-3 69-4 69-5 69-7 69-8 69-9 69-11 69-14 69-15 69-16 69-17 69-18 69-19 69-20 69-21 69-22
Preface
Page xii
70 GEN ROUTINES
70.1 70.2 70.3 70.4 70.5 70.6 70.7 70.8 70.9 70.10 70.11 70.12 70.13 70.14 70.15 70.16 70.17 70.18 70.19 GEN01 GEN02 GEN03 GEN04 GEN05, GEN06 GEN08 GEN09, GEN11 GEN12 GEN13, GEN15 GEN16 GEN17 GEN20 GEN21 GEN23 GEN25, GEN28
70-1
70-2 70-4 70-5 70-6 70-7 70-8 70-9 70-10 70-11 70-12 70-13 70-15 70-16 70-17 70-18 70-20 70-21 70-22 70-23
GEN07 GEN10, GEN19 GEN14
GEN27
71 THE CSOUND COMMAND

71.1 71.2 71.3 71.4 71.5 Order of Precedence Generic Flags PC Windows Specific flags Macintosh Specific Flags Description
71-1
71-1 71-1 71-2 71-4 71-4
72 UNIFIED FILE FORMAT FOR ORCHESTRAS AND SCORES

72.1 72.2 72.3 72.4 Description Structured Data File Format Example Command Line Parameter File
72-1
72-1 72-1 72-2 72-3
73 SCORE FILE PREPROCESSING

73.1 73.2 The Extract Feature Independent Pre-Processing with Scsort
73.1
73.1 73.2
74 UTILITY PROGRAMS
74.1 74.2 74.3 74.4 74.5 74.6 74.7 sndinfo hetro lpanal pvanal cvanal pvlook sdif2ads
74-1
74-3 74-4 74-6 74-8 74-10 74-11 74-15
Preface
Page xiii
75 CSCORE
75.1 75.2 75.3 75.4 Events, Lists, and Operations Writing a Main Program More Advanced Examples Compiling a Cscore Program
75.1
75.2 75.4 75.9 75.11
76 ADDING YOUR OWN CMODULES TO CSOUND 77 APPENDIX A: MISCELLANEOUS INFORMATION

77.1 77.2 77.3 77.4 77.5 77.6 77.7 Pitch Conversion Sound Intensity Values (for a 1000 Hz tone) Formant Values Window Functions SoundFont2 File Format Print Edition Update Procedure Manual Update History
76-1 771
771 773 774 775 779 7710 7711
Preface
Page xiv
Finder
Symbols
- 16-1 != 12-1 #define(orc) 13-1 #define(sco) 69-5 #include(orc) 13-3 #include(sco) 69-7 #undef(orc) 13-1 #undef(sco) 69-5 $NAME(orc) 13-1 $NAME(sco) 69-5 % 16-1 && 16-1 ( 69-4 ) 69-4 * 16-1 / 16-1 ? 12-1 @ 69-8 @@ 69-8 ^ 16-1 { 69-4 || 16-1 } 69-4 ~ 69-4 +(orc) 16-1 <(orc) 12-1 <(sco) 69-4 <= 12-1 = 5-1 == 12-1 >(orc) 12-1 >= 12-1
A
a Statement 69-14 abs 17-1 active 8-1 adsr 32-3 adsyn 37-1 adsynt 37-3 aftouch 24-1 alpass 56-1 ampdb 19-1 ampdbfs 19-1 ampmidi 24-1 areson 50-1 aresonk 50-1 atone 50-1 atonek 50-1 atonex 50-3
convolve 59-1 cos 18-1 cosh 18-1 cosinv 18-1 cps2pch 23-1 cpsmidi 24-1 cpsmidib 24-1 cpsoct 22-1 cpspch 22-1 cpstmid 24-3 cpsxpch 23-1 cpuprc 8-2 cross2 59-4 crunch 43-12 ctrl14 25-2 ctrl21 25-2 ctrl7 25-2 ctrlinit 3-5 cvanal 74-10
expsegr 32-1
F
f Statement 69-9 filelen 67-1 filenchnls 67-1 filepeak 67-1 filesr 67-1 filter2 50-20 fin 66-5 fini 66-5 fink 66-5 fiopen 66-3 flanger 58-3 fmb3 38-3 fmbell 38-3 fmmetal 38-3 fmpercfl 38-3 fmrhode 38-3 fmvoice 38-2 fmwurlie 38-3 fof 40-1 fof2 40-1 fog 40-3 fold 61-4 follow 11-7 follow2 11-11 foscil 38-1 foscili 38-1 fout 66-3 fouti 66-3 foutir 66-3 foutk 66-3 frac 17-1 ftgen 3-4 ftlen 47-1 ftlptim 47-1 ftsr 47-1
B
b Statement 69-16 babo 56-7 balance 53-1 bamboo 43-14 betarand 46-2 bexprnd 46-2 biquad 50-12 birnd 20-1 bug reports, code 19 bug reports, documentation 77 10 butbp 50-19 butbr 50-19 buthp 50-19 butlp 50-19 butterbp 50-19 butterbr 50-19 butterhp 50-19 butterlp 50-19 button 11-13 buzz 36-1
D
dam 53-2 dbamp 19-1 dbfsamp 19-1 dcblock 51-5 delay 55-1 delay1 55-1 delayr 55-1 delayw 55-1 deltap 55-3 deltap3 55-3 deltapi 55-3 deltapn 55-3 diff 61-1 diskin 64-1 dispfft 68-1 display 68-1 distort1 58-4 divz 5-1 downsamp 61-1 dripwater 43-14 dumpk 66-1 dumpk2 66-1 dumpk3 66-1 dumpk4 66-1
Tags, Files and Extensions

.csd 72-1 .csoundrc 72-3 .orc 11 .sco 11 <CsInstruments> 72-1 <CsMidifileB> 72-1 <CsOptions> 72-1 <CsoundSynthesizer> 72-1 <CsSampleB> 72-2 <CsScore> 72-1 <CsVersion> 72-2 csound.txt 2-2 CSSTRNGS 2-2 INCDIR 2-2 SADIR 2-2 SFDIR 2-2 SSDIR 2-2
G
gain 53-1 gauss 46-2 gbuzz 36-1 GEN01 70-2 GEN02 70-4 GEN03 70-5 GEN04 70-6 GEN05 70-7 GEN06 70-8 GEN07 70-7 GEN08 70-9 GEN09 70-10 GEN10 70-10 GEN11 70-11 GEN12 70-12 GEN13 70-13
Preface Page xv
C
cabasa 43-12 cauchy 46-2 chanctrl 25-4 checkbox 11-13 clear 66-6 clip 53-3 clockoff 10-1 clockon 10-1 comb 56-1 control 11-12 convle 59-1
E
e Statement 69-19 endin 4-1 envlpx 52-1 envlpxr 52-1 exp 17-1 expon 32-1 exprand 46-2 expseg 32-1 expsega 32-1
GEN14 70-13 GEN15 70-15 GEN16 70-16 GEN17 70-17 GEN19 70-10 GEN20 70-18 GEN21 70-20 GEN23 70-21 GEN25 70-22 GEN27 70-22 GEN28 70-23 gogobel 43-6 goto 14-1 grain 40-5 granule 40-7 guiro 43-14
H
harmon 58-1 hetro 74-4 hilbert 50-16 hrtfer 60-9 hsboscil 37-5
line 32-1 linen 52-1 linenr 52-1 linrand 46-2 linseg 32-1 linsegr 32-1 locsend 60-3 locsig 60-3 log 17-1 log10 17-1 logbtwo 17-2 lorenz 43-8 loscil 39-1 loscil3 39-1 lowpass2 50-10 lowres 50-8 lowresx 50-8 lpanal 74-6 lpf18 50-22 lpfreson 45-1 lpinterp 45-3 lposcil 39-3 lposcil3 39-3 lpread 45-1 lpreson 45-1 lpslot 45-3
N
n Statement 69-22 nchnls 3-1 nestedap 56-5 nlfilt 51-1 noise 46-6 noteoff 28-1 noteon 28-1 noteondur 28-1 noteondur2 28-1 notnum 24-1 np 69-4 nreverb 56-3 nrpn 29-3 nsamp 47-1 ntrpol 61-3
O
octcps 22-1 octmidi 24-1 octmidib 24-1 octpch 22-1 oscil 35-1 oscil1 33-1 oscil1i 33-1 oscil3 35-1 oscili 35-1 osciln 33-1 out 65-1 out32 65-3 outc 65-3 outch 65-3 outh 65-1 outiat 29-1 outic 29-1 outic14 29-1 outipat 29-1 outipb 29-1 outipc 29-1 outkat 29-1 outkc 29-1 outkc14 29-1 outkpat 29-1 outkpb 29-1 outkpc 29-1 outo 65-1 outq 65-1 outq1 65-1 outq2 65-1 outq3 65-1 outq4 65-1 outs 65-1 outs1 65-1 outs2 65-1 outx 65-3 outz 65-3
I
i 17-1 i Statement 69-11 if 14-1 igoto 14-1 ihold 7-1 in 64-1 in32 64-3 inch 64-3 inh 64-1 init 5-1 initc14 25-1 initc21 25-1 initc7 25-1 ino 64-1 inq 64-1 ins 64-1 instr 4-1 int 17-1 integ 61-1 interp 61-1 inx 64-3 inz 64-3
M
m Statement 69-21 mac 21-5 maca 21-5 macros, orchestra 13-1 madsr 32-3 mandol 43-5 marimba 43-3 massign 3-5 maxalloc 8-2 mclock 30-1 mdelay 29-4 MIDI sliders 26-1 midic14 25-2 midic21 25-2 midic7 25-2 midictrl 24-1 midiin 27-1 midion 28-3 midion2 28-4 midiout 27-2 mirror 54-1 moog 43-1 moogvcf 50-12 moscil 28-3 mpulse 36-5 mrtmsg 30-1 multiple files, orchestra 13-3 multiple files, score 697 multitap 55-5 mxadsr 32-3
pcauchy 46-2 pchbend 24-1 pchmidi 24-1 pchmidib 24-1 pchoct 22-1 peak 11-9 phaser1 58-6 phaser2 58-6 phasor 34-1 pinkish 46-4 pitch 11-1 pitchamdf 11-3 planet 43-10 pluck 42-1 poisson 46-2 port 50-1 portk 50-1 poscil 35-2 poscil3 35-2 pow 21-3 powoftwo 17-2 pp 69-4 prealloc 8-2 print 68-1 printk 68-2 printk2 68-4 printks 68-2 product 21-2 pset 3-2 pvadd 44-6 pvanal 74-8 pvbufread 44-3 pvcross 44-3 pvinterp 44-3 pvlook 74-11 pvoc 44-1 pvread 44-3
R
r Statement 69-20 rand 46-1 randh 46-1 randi 46-1 readclock 10-1 readk 66-1 readk2 66-1 readk3 66-1 readk4 66-1 reinit 15-1 release 31-1 repluck 42-4 reson 50-1 resonk 50-1 resonr 50-4 resonx 50-3 resony 50-7 resonz 50-4 reverb 56-1 reverb2 56-3 rezzy 50-12 rigoto 15-1 rireturn 15-1 rms 53-1 rnd 20-1
Preface Page xvi
K
kgoto 14-1 kr 3-1 ksmps 3-1
L
lfo 35-3 limit 54-1
P
pan 60-1 pareq 51-3
S
s Statement 69-18 s16b14 26-1 s32b14 26-1 samphold 61-1 sandpaper 43-12 scans 41-5 scanu 41-3 schedkwhen 6-3 schedule 6-1 schedwhen 6-1 sdif2ads 74-15 seed 3-3 sekere 43-12 sensekey 11-14 setctrl 11-12 sfilist 39-4 sfinstr 39-4 sfinstrm 39-4 sfload 39-4 sfpassign 39-4 sfplay 39-4 sfplaym 39-4 sfplist 39-4 sfpreset 39-4 shaker 43-2 sin 18-1 sinh 18-1 sininv 18-1 sleighbells 43-14 slider16 26-1 slider16f 26-1 slider32 26-1 slider32f 26-1 slider64 26-1 slider64f 26-1 slider8 26-1 slider8f 26-1 sndinfo 74-3 sndwarp 40-10 sndwarpst 40-10 soundin 64-1 soundout 65-1 soundouts 65-1 space 60-5 spdist 60-5
specaddm 63-2 specdiff 63-2 specdisp 63-5 specfilt 63-2 spechist 63-2 specptrk 63-3 specscal 63-2 specsum 63-5 spectrum 63-6 spsend 60-5 sqrt 17-1 sr 3-1 stix 43-12 streson 57-3 strset 3-2 sum 21-1 svfilter 50-14
tempo 11-10 tigoto 14-1 timeinstk 9-1 timeinsts 9-1 timek 9-1 times 9-1 timout 14-1 tival 5-1 tone 50-1 tonek 50-1 tonex 50-3 transeg 32-4 trigger 11-8 trirand 46-2 turnoff 7-1 turnon 6-4
W
weibull 46-2 wgbow 42-5 wgbowedbar 42-9 wgbrass 42-7 wgclar 42-8 wgflute 42-6 wgpluck 42-3 wgpluck2 42-4 wguide1 57-1 wguide2 57-1 wrap 54-1
X
xadsr 32-3 x-class noise generators 46-2 xtratim 31-1 xyin 11-10
T
t Statement 69-15 table 33-1 table3 33-1 tablecopy 49-3 tablegpw 49-3 tablei 33-1 tableicopy 49-3 tableigpw 49-3 tableikt 48-1 tableimix 49-3 tableiw 49-1 tablekt 48-1 tablemix 49-3 tableng 47-2 tablera 49-5 tableseg 44-3 tablew 49-1 tablewa 49-5 tablewkt 49-1 tablexseg 44-3 tambourine 43-14 tan 18-1 tanh 18-1 taninv 18-1 taninv2 21-4 tbvcf 50-23 tempest 11-5
U
unirand 46-2 upsamp 61-1
V
v Statement 69-17 vbap16 60-10 vbap16move 60-10 vbap4 60-10 vbap4move 60-10 vbap8 60-10 vbap8move 60-10 vbaplsinit 60-10 vbapz 60-10 vbapzmove 60-10 vco 36-3 vdelay 55-6 vdelay3 55-6 veloc 24-1 vibes 43-3 vincr 66-6 vlowres 50-9 voice 43-7 vpvoc 44-1
Z
zacl 62-6 zakinit 62-2 zamod 62-6 zar 62-5 zarg 62-5 zaw 62-3 zawm 62-3 zfilter2 50-20 zir 62-5 ziw 62-3 ziwm 62-3 zkcl 62-6 zkmod 62-6 zkr 62-5 zkw 62-3 zkwm 62-3
Preface
Page xvii
Preface
Page xviii
PREFACE
by Barry Vercoe, MIT Media Lab
Realizing music by digital computer involves synthesizing audio signals with discrete points or samples representative of continuous waveforms. There are many ways to do this, each affording a different manner of control. Direct synthesis generates waveforms by sampling a stored function representing a single cycle; additive synthesis generates the many partials of a complex tone, each with its own loudness envelope; subtractive synthesis begins with a complex tone and filters it. Non-linear synthesis uses frequency modulation and waveshaping to give simple signals complex characteristics, while sampling and storage of a natural sound allows it to be used at will. Since comprehensive moment-by-moment specification of sound can be tedious, control is gained in two ways: 1) from the instruments in an orchestra, and 2) from the events within a score. An orchestra is really a computer program that can produce sound, while a score is a body of data which that program can react to. Whether a rise-time characteristic is a fixed constant in an instrument, or a variable of each note in the score, depends on how the user wants to control it. The instruments in a Csound orchestra (.orc) are defined in a simple syntax that invokes complex audio processing routines. A score (.sco) passed to this orchestra contains numerically coded pitch and control information, in standard numeric score format. Although many users are content with this format, higher level score processing languages are often convenient. The programs making up the Csound system have a long history of development, beginning with the Music 4 program written at Bell Telephone Laboratories in the early 1960s by Max Mathews. That initiated the stored table concept and much of the terminology that has since enabled computer music researchers to communicate. Valuable additions were made at Princeton by the late Godfrey Winham in Music 4B; my own Music 360 (1968) was very indebted to his work. With Music 11 (1973) I took a different tack: the two distinct networks of control and audio signal processing stemmed from my intensive involvement in the preceding years in hardware synthesizer concepts and design. This division has been retained in Csound. Because it is written entirely in C, Csound is easily installed on any machine running Unix or C. At MIT it runs on VAX/DECstations under Ultrix 4.2, on SUNs under OS 4.1, SGIs under 5.0, on IBM PCs under DOS 6.2 and Windows 3.1, and on the Apple Macintosh under ThinkC 5.0. With this single language for defining the audio signal processing, and portable audio formats like AIFF and WAV, users can move easily from machine to machine. The 1991 version added phase vocoder, FOF, and spectral data types. 1992 saw MIDI converter and control units, enabling Csound to be run from MIDI score-files and external keyboards. In 1994 the sound analysis programs (lpc, pvoc) were integrated into the main load module, enabling all Csound processing to be run from a single executable, and Cscore could pass scores
Preface
Page 11
directly to the orchestra for iterative performance. The 1995 release introduced an expanded MIDI set with MIDI-based linseg, butterworth filters, granular synthesis, and an improved spectral-based pitch tracker. Of special importance was the addition of run-time event generating tools (Cscore and MIDI) allowing run-time sensing and response setups that enable interactive composition and experiment. It appeared that real-time software synthesis was now showing some real promise.
Preface
Page 12
1.1
Where to Get Public Csound and the Csound Manual

Public Csound is available for download via anonymous ftp from : ftp://ftp.maths.bath.ac.uk/pub/dream or ftp://ftp.musique.umontreal.ca/pub/mirrors/dream The Acrobat Edition and HTML Edition of this manual is available for browser download from: http://www.lakewoodsound.com/csound or via anonymous ftp from: ftp://ftp.csounds.com/manual
Preface
Page 13
1.2
How to Install Csound
MACINTOSH
Detailed instructions for installing and configuring Csound on Macintosh systems may be obtained from: http://mitpress.mit.edu/e-books/csound/fpage/gs/mac/a/a.html
W I N DO W S 95/98
Detailed instructions for installing and configuring Csound on Windows 95 or Windows 98 systems may be obtained from: http://mitpress.mit.edu/e-books/csound/fpage/gs/pc/pc.html
MS-DOS
A N D W I N DO W S 3. X Detailed instructions for installing and configuring Csound on MS-DOS or Windows 3.x systems may be obtained from: http://hem.passagen.se/rasmuse/PCinstal.htm
L I N U X (D E V E L O P E R S V E R S I O N )
Introduction to the Developers Linux Version Building Csound for UNIX and Linux machines has been possible thanks to John Fitchs Csound.tar.gz source file kept at: ftp://ftp.maths.bath.ac.uk/pub/dream/newest
This source tree builds Csound on a variety of UNIX-type systems, including the NeXT, Suns Solaris, SGI machines, and Intel-based Linux. It should be noted that John also maintains a Linux binary at the Bath repository. That version is built from his canonical sources. In 1998 a group of developers prepared a new version of Csound for Linux. This version (often referred to as the "unofficial" distribution) aims to deliver a modern package for Linux users. It offers a variety of amenities specific to Linux systems, including these items: Enhanced makefile system autoconf and configure supported for site-specific build Support for Jaroslav Kyselas ALSA sound drivers Support for 64-bit Alpha systems Full MIDI and real-time audio support Builds shared library (libcsound.so) for greatly reduced memory footprint Includes Robin Whittles random number generator Provided in various popular Linux distribution packaging formats Utilizes .csoundrc resource file Provides a high-priority scheduler for improved real-time i/o Includes support for full-duplex under the OSS/Free and OSS/Linux drivers CVS and bug-tracking system established for developers This distributions code base originates with the sources provided by John Fitch at the Bath site. Every effort is made to ensure compatibility with those sources at the opcode level, and users should have no trouble running most orc/sco files or .csd files made for Csound on other operating systems. The makefile structure has been provided by Nicola Bernardini. He also maintains the CVS repository. Other features have been added by developers Ed Hall (Alpha port), Fred Floberg (scheduler), Robin Whittle, and Steve Kersten (full-duplex under OSS driver). RPM and DEB packages are sporadically available from Damien Miller and Guenter Geiger.
Preface
Page 14
Building the developers version is quite simple, using the familiar ./configure; make depend; make; make install command sequence. Instructions for compiling and installing Csound are provided with the package, along with other relevant documentation. A maillist has been established for developers and users of this package, and a bug-tracking system has been set up by Damien Miller. Preparing Linux Audio for Csound As long as the basic Linux audio system is properly configured and installed, no special efforts need to be made in order to enjoy audio output from sound. The default real-time audio output device (devaudio) is defined as /dev/dsp in Csound itself, although other audio devices (/dev/audio, /dev/dspW) can be specified if so desired. Using the Developers Version This version is designed to be opcode-compatible with any other version of Csound. However, some new options have been added which may require clarification. Real-time audio output can be as simple as this: csound -o devaudio -V 75 my.orc my.sco The -V flag is a Linux-specific output volume control from Jonathan Mohr. Note that it will work only with the OSS/Free and OSS/Linux drivers. Here we get a little more complicated: csound --sched --ossin=/dev/dsp0 --ossout=/dev/dsp1 my.* This example invokes Fred Flobergs high-priority scheduler (which will automatically disable graphics output) and Steve Kerstens support for full-duplex using either the OSS/Free driver included with the Linux kernel or the commercially available OSS/Linux driver. Linux users can use the asterisk as a wildcard for the orc/sco extensions. However, if you have my.orc, my.sco, and my.txt within the same directory the compiler will get confused and the wildcard wont work. If more than one soundcard is present in the system, ALSA users have the option of choosing which card will function for either audio input or output. The command sequence then appears so: csound --incard=1 --outcard=2 my.orc your.sco The standard advice regarding audio buffer settings holds true for Linux as well as for any other version. If the audio output is choppy you may need to adjust the value for the b flag which controls the sample frame size for the software audio buffer. The best setting will depend upon various aspects of your machine system, including CPU speed, memory limits, hard-disk performance, etc. Supported options for MIDI include the -Q (MIDI output device) and -K (MIDI input port) flags from Gabriel Maldonados DirectCsound. Here is an example which uses one of Gabriels opcodes, requiring the use of a MIDI output port: csound -Q0 -n my_moscil.orc my_moscil.sco The -Q0 flag selects the first available MIDI output device, -n cancels writing the output to disk.
Preface
Page 15
It should be noted that, for Linux at least, in the opcode for this instrument (moscil) the sample rate determines the tempo of events. Setting the control rate (kr) to equal the sample rate (sr), sr=kr a higher sample rate will result in a slower performance. When sr=390000 (yes, you read that correctly, its a sample rate of three hundred and ninety thousand) then the MIDI event performance output is approximately 60 BPM (beats per minute). At that sample rate a score tempo statement of t 0 60will actually mean 60 bpm. In essence, the sample rate acts as a restraint or throttle on the tempo of the MIDI event stream. Using MIDI for real-time input is simple: csound --sched -o devaudio -M/dev/midi my_midi_in.* With correctly written orc/sco files this example will allow real-time control of Csound via whatever controlling device is hooked up to /dev/midi. If more than one MIDI device is present in the machine the user can specify which to use: csound --sched -o devaudio -M/dev/midi01 my_midi_in.* That sequence will select the second MIDI device for MIDI input. Here we use a Type 0 standard MIDI file for the controlling input: csound --sched -o devaudio -T -F/home/midfiles/my_type_0.mid my_cool.orc my_cool.sco In these last two examples the score file provides only the necessary function tables and a place-holder to indicate how long Csound should stay active: f1 0 8192 10 1 f0 240 e ; a sine wave ; stay active for 240 seconds
However, the -T flag will halt performance as soon as the end of the MIDI file is reached. Availability The Linux developers version of Csound is available in source and binary distributions. The main distribution sites are at AIMI in Italy: http://AIMI.dist.unige.it/AIMICSOUND/AIMICSOUND_home.html and the ftp server for the Music Technology Department at Bowling Green State University in the USA: ftp://mustec.bgsu.edu/pub/linux
Developer Maurizio Umberto Puxeddu has also established a distribution point, though at this time it is version-specific and not browsable. For more information regarding his site, and for more information generally regarding Linux Csound, see this Web page: http://www.bright.net/~dlphilp/linux_csound.html Credits First thanks go to Barry Vercoe for creating Csound and allowing it to be freely and publicly distributed and to John Fitch for maintaining the canonical source packages (including his own build for Linux). Special thanks go to the following persons for their development assistance and/or spiritual guidance: Paul Barton-Davis Nicola Bernardini Richard Boulanger Fred Floberg Ed Hall Steve Kersten
Preface Page 16
Gabriel Maldonado Damien Miller Maurizio Umberto Puxeddu Larry Troxler Robin Whittle
My apologies to anyone Ive left out. Please send corrections and emendations of this document to me at my email address below. Dave Phillips dlphilp@bright.net September 1999
OTHER PLATFORMS
For information on availability of Csound for other platforms, see The Csound Frontpage: http://mitpress.mit.edu/e-books/csound/frontpage.html
Preface
Page 17
1.3
How to use the Csound Manual
The Csound Manual is arranged as a Reference manual (not a tutorial), since that is the form the user will eventually find most helpful when inventing instruments. Csound can be a demanding experience at first. Hence it is highly advisable to peruse the tutorials included in the Supplement to this Manual. Once the basic concepts are grasped from the beginning tutorial, the reader might let himself into the remainder of the text by locating the information presented in the Reference entries that follow.
Preface
Page 18
1.4
The Csound Mailing List
A Csound Mailing List exists to discuss Csound. It is run by John ffitch of Bath University, UK. To have your name put on the mailing list send an empty message to: csound-subscribe@lists.bath.ac.uk Posts sent to csound@lists.bath.ac.uk go to all subscribed members of the list.
B U G R E PO R T S
Suspected bugs in the code may be submitted to the list.
Preface
Page 19
This page intentionally left blank
Preface
Page 110
SYNTAX
OF THE
ORCHESTRA
An orchestra statement in Csound has the format: label: result argument1, argument2, ... opcode ;comments The label is optional and identifies the basic statement that follows as the potential target of a go-to operation (see Program Control Statements). A label has no effect on the statement per se. Comments are optional and are for the purpose of letting the user document his orchestra code. Comments always begin with a semicolon (;) and extend to the end of the line. The remainder (result, opcode, and arguments) form the basic statement. This also is optional, i.e. a line may have only a label or comment or be entirely blank. If present, the basic statement must be complete on one line, and is terminated by a carriage return and line feed. Occasionally in this manual, a statement is divided between two lines. This is for printing convenience only, and does not apply to the HTML Edition. In orchestra files, statements must be complete on one line, without a carriage return or line feed before the end of the statement. The opcode determines the operation to be performed; it usually takes some number of input values (or arguments, with a maximum value of about 800); and it usually has a result field variable to which it sends output values at some fixed rate. There are four possible rates: once only, at orchestra setup time (effectively a permanent assignment); once at the beginning of each note (at initialization (init) time: i-rate); once every performance-time control loop (perf-time control rate, or k-rate); once each sound sample of every control loop (perf-time audio rate, or a-rate).
Syntax of the Orchestra
Page 2-1
2.1
Directories and Files

Many generators and the Csound command itself specify filenames to be read from or written to. These are optionally full pathnames, whose target directory is fully specified. When not a full path, filenames are sought in several directories in order, depending on their type and on the setting of certain environment variables. The latter are optional, but they can serve to partition and organize the directories so that source files can be shared rather than duplicated in several user directories. The environment variables can define directories for soundfiles SFDIR, sound samples SSDIR, sound analysis SADIR, and include files for orchestra and score files INCDIR. The search order is: 1. Soundfiles being written are placed in SFDIR (if it exists), else the current directory. 2. Soundfiles for reading are sought in the current directory, then SSDIR, then SFDIR. 3. Analysis control files for reading are sought in the current directory, then SADIR. 4. Files of code to be included in orchestra and score files (with #include) are sought first in the current directory, then in the same directory as the orchestra or score file (as appropriate), then finally INCDIR. Beginning with Csound version 3.54, the file csound.txt contains the messages (in binary format) that Csound uses to provide information to the user during performance. This allows for the messages to be in any language, although the default is English. This file must be placed in the same directory as the Csound executable. Alternatively, this file may be stored in SFDIR, SSDIR, or SADIR. Unix users may also keep this file in usr/local/lib/. The environment variable CSSTRNGS may be used to define the directory in which the database resides. This can be overridden with the -j command line option. (New in version 3.55)
Page 2-2
2.2
Nomenclature
Throughout this document, opcodes are indicated in boldface and their argument and result mnemonics, when mentioned in the text, are given in italics. Argument names are generally mnemonic (amp, phs), and the result is usually denoted by the letter r. Both are preceded by a type qualifier i, k, a, or x (e.g. kamp, iphs, ar). The prefix i denotes scalar values valid at note init time; prefixes k or a denote control (scalar) and audio (vector) values, modified and referenced continuously throughout performance (i.e. at every control period while the instrument is active). Arguments are used at the prefix-listed times; results are created at their listed times, then remain available for use as inputs elsewhere. With few exceptions, argument rates may not exceed the rate of the result. The validity of inputs is defined by the following: arguments with prefix i must be valid at init time; arguments with prefix k can be either control or init values (which remain valid); arguments with prefix a must be vector inputs; arguments with prefix x may be either vector or scalar (the compiler will distinguish). All arguments, unless otherwise stated, can be expressions whose results conform to the above. Most opcodes (such as linen and oscil) can be used in more than one mode, which one being determined by the prefix of the result symbol. Thoughout this manual, the term opcode is used to indicate a command that usually produces an a-, k-, or i-rate output, and always forms the basis of a complete Csound orchestra statement. Items such as + or sin(x) or, ( a >= b ? c : d) are called operators. In the Csound orchestra, statements fall into twelve major categores, consisting of sixtyfive sub-categories. Each is in a separate chapter of this manual. The categories (and corresponding chapter numbers) are as follows:
Orchestra Syntax 3: Orchestra Header Statements 4: Instrument Block Statements 5: Variable Initialization Instrument Control 6: Instrument Invocation 7: Duration Control Statements 8: Real-time Performance Control 9: Time Reading 10: Clock Control 11: Sensing and Control 12: Conditional Values 13: Macros 14: Program Flow Control 15: Reinitialization
Page 2-3
Mathematical Operations 16: Arithmetic and Logic Operations 17: Mathematical Functions 18: Trigonometric Functions 19: Amplitude Functions 20: Random Functions 21: Opcode Equivalents of Functions Pitch Converters 22: Functions 23: Tuning Opcodes MIDI Support 24: Converters 25: Controller Input 26: Slider Banks 27: Generic I/O 28: Note-on/Note-off 29: MIDI Message Output 30: Real-time Messages 31: MIDI Event Extenders Signal Generators 32: Linear and Exponential Generators 33: Table Access 34: Phasors 35: Basic Oscillators 36: Dynamic Spectrum Oscillators 37: Additive Synthesis/Resynthesis 38: FM Synthesis 39: Sample Playback 40: Granular Synthesis 41: Waveguide Physical Modeling 42: Models and Emulations 43: STFT Resynthesis (Vocoding) 44: LPC Resynthesis 45: Random (Noise) Generators Function Table Control 46: Tables Queries 47: Table Selection 48: Read/Write Operations
The Public Csound Reference Manual Version 4.10 Syntax of the Orchestra Page 2-4
Signal Modifiers 49: Standard Filters 50: Specialized Filters 51: Envelope Modifiers 52: Amplitude Modifiers 53: Signal Limiters 54: Delay 55: Reverberation 56: Waveguides 57: Special Effects 58: Convolution and Morphing 59: Panning and Spatialization 60: Sample Level Operators Zak Patch System 61: Zak Patch System Operations Using Spectral Data Types 62: Operations Using Spectral Data Types Signal Input and Output 63: Input 64: Output 65: File I/O 66: Sound File Queries 67: Printing and Display
Page 2-5
2.3
Orchestra Statement Types

An orchestra program in Csound is comprised of orchestra header statements which set various global parameters, followed by a number of instrument blocks representing different instrument types. An instrument block, in turn, is comprised of ordinary statements that set values, control the logical flow, or invoke the various signal processing subroutines that lead to audio output. An orchestra header statement operates once only, at orchestra setup time. It is most commonly an assignment of some value to a global reserved symbol , e.g. sr = 20000. All orchestra header statements belong to a pseudo instrument 0, an init pass of which is run prior to all other instruments at score time 0. Any ordinary statement can serve as an orchestra header statement, e.g. gifreq = cpspch(8.09) provided it is an init-time only operation. An ordinary statement runs at either init time or performance time or both. Operations which produce a result formally run at the rate of that result (that is, at init time for i-rate results; at performance time for k- and a-rate results), with the sole exception of the init opcode. Most generators and modifiers, however, produce signals that depend not only on the instantaneous value of their arguments but also on some preserved internal state. These performance-time units therefore have an implicit init-time component to set up that state. The run time of an operation which produces no result is apparent in the opcode. Arguments are values that are sent to an operation. Most arguments will accept arithmetic expressions composed of constants, variables, reserved symbols, value converters, arithmetic operations, and conditional values.
Page 2-6
2.4
Constants and Variables

constants are floating point numbers, such as 1, 3.14159, or -73.45. They are available continuously and do not change in value. variables are named cells containing numbers. They are available continuously and may be updated at one of the four update rates (setup only, i-rate, k-rate, or a-rate). i- and k-rate variables are scalars (i.e. they take on only one value at any given time) and are primarily used to store and recall controlling data, that is, data that changes at the note rate (for irate variables) or at the control rate (for k-rate variables). i- and k-variables are therefore useful for storing note parameter values, pitches, durations, slow-moving frequencies, vibratos, etc. a-rate variables, on the other hand, are arrays or vectors of information. Though renewed on the same perf-time control pass as k-rate variables, these array cells represent a finer resolution of time by dividing the control period into sample periods (see ksmps). a-rate variables are used to store and recall data changing at the audio sampling rate (e.g. output signals of oscillators, filters, etc.). A further distinction is that between local and global variables. local variables are private to a particular instrument, and cannot be read from or written into by any other instrument. Their values are preserved, and they may carry information from pass to pass (e.g. from initialization time to performance time) within a single instrument. Local variable names begin with the letter p, i, k, or a. The same local variable name may appear in two or more different instrument blocks without conflict. global variables are cells that are accessible by all instruments. The names are either like local names preceded by the letter g, or are special reserved symbols. Global variables are used for broadcasting general values, for communicating between instruments (semaphores), or for sending sound from one instrument to another (e.g. mixing prior to reverberation). Given these distinctions, there are eight forms of local and global variables: Type reserved symbols score parameter fields v-set symbols init variables MIDI controllers control signals audio signals spectral data types When Renewable permanent i-time i-time i-time any time p-time, k-rate p-time, a-rate k-rate Local -pnumber vnumber iname cnumber kname aname wname Global rsymbol -gvnumber gv giname gi -gkname gk ganame ga --
where rsymbol is a special reserved symbol (e.g. sr, kr), number is a positive integer referring to a score pfield or sequence number, and name is a string of letters and/or digits with local or global meaning. As might be apparent, score parameters are local i-rate variables whose values are copied from the invoking score statement just prior to the init pass through an instrument, while MIDI controllers are variables which can be updated asynchronously from a MIDI file or MIDI device.
Page 2-7
2.5
Expressions
Expressions may be composed to any depth. Each part of an expression is evaluated at its own proper rate. For instance, if the terms within a sub-expression all change at the control rate or slower, the sub-expression will be evaluated only at the control rate; that result might then be used in an audio-rate evaluation. For example, in k1 + abs(int(p5) + frac(p5) * 100/12 + sqrt(k1))
the 100/12 would be evaluated at orch init, the p5 expressions evaluated at note i-time, and the remainder of the expression evaluated every k-period. The whole might occur in a unit generator argument position, or be part of an assignment statement.
Page 2-8
ORCHESTRA SYNTAX: ORCHESTRA HEADER STATEMENTS

sr, kr, ksmps, nchnls
= = = = iarg iarg iarg iarg
3.1
sr kr ksmps nchnls
DESCRIPTION
These statements are global value assignments, made at the beginning of an orchestra, before any instrument block is defined. Their function is to set certain reserved symbol variables that are required for performance. Once set, these reserved symbols can be used in expressions anywhere in the orchestra. sr = (optional) set sampling rate to iarg samples per second per channel. The default value is 44100. kr = (optional) set control rate to iarg samples per second. The default value is 4410. ksmps = (optional) set the number of samples in a control period to. This value must equal sr/kr. The default value is 10. nchnls = (optional) set number of channels of audio output to iarg. (1 = mono, 2 = stereo, 4 = quadraphonic.) The default value is 1 (mono). In addition, any global variable can be initialized by an init-time assignment anywhere before the first instr statement. All of the above assignments are run as instrument 0 (ipass only) at the start of real performance. Beginning with Csound version 3.46, either sr, kr, or ksmps may be omitted. Csound will attempt to calculate the omitted value from the specified values, but it should evaluate to an integer.
E X A M PL E
sr = 10000 kr = 500 ksmps = 20 gi1 = sr/2. ga init 0 itranspose = octpch(.0l)
Orchestra Syntax: Orchestra Header Statements
Page 3-1
3.2
strset, pset
strset pset iarg, stringtext con1, con2, con3,...
DESCRIPTION
Allow certain global parameters to be initialized at orchestra load time, rather than instrument initialization or performance time.
I N I T I A L I ZA T I O N
iarg numeric value to be associated with an alphanumeric string con1, con2, etc. preset values for a MIDI instrument strset (optional) allows a string, such as a filename, to be linked with a numeric value. Its use is optional. pset (optional) defines and initializes numeric arrays at orchestra load time. It may be used as an orchestra header statement (i.e. instrument 0) or within an instrument. When defined within an instrument, it is not part of its i-time or performance operation, and only one statement is allowed per instrument. These values are available as i-time defaults. When an instrument is triggered from MIDI it only gets p1 and p2 from the event, and p3, p4, etc. will receive the actual preset values.
E X A M PL E S
The following statement, used in the orchestra header, will allow the numeric value 10 to substituted anywhere the soundfile asound.wav is called for.
strset 10, asound.wav
The example below illustrates pset as used within an instrument.

instr 1 pset oscil
a1
0,0,3,4,5,6 10000, 440, p6
; pfield substitutes
Page 3-2
3.3
seed
seed ival
DESCRIPTION
Sets the global seed value for all x-class noise generators, as well as other opcodes that use a random call, such as grain. rand, randi, randh, rnd(x), and birnd(x) are not affected by seed.
ival value to be used as the random generator(s) seed value
PERFORMANCE
Use of seed will provide predictable results from an orchestra using with random generators, when required from multiple performances. When specifying a seed value, ival should be an integer between 0 and 232. If ival = 0, the value of ival will be derived from the system clock.
Page 3-3
3.4
gir
ftgen
ftgen ifn, itime, isize, igen, iarga[, iargb...iargz]
DESCRIPTION
Generate a score function table from within the orchestra.
gir either a requested or automatically assigned table number above 100. If used within an instrument, may be local variable ir. ifn requested table number If ifn is zero, the number is assigned automatically and the value placed in gir. Any other value is used as the table number itime is ignored, but otherwise corresponds to p2 in the score f statement. isize table size. Corresponds to p3 of the score f statement. igen function table GEN routine. Corresponds to p4 of the score f statement. iarga-iargz function table arguments. Correspond to p5 through pn of the score f statement.
PERFORMANCE
This is equivalent to table generation in the score with the f statement.
AUTHOR
Barry Vercoe MIT, Cambridge, Mass 1997
Page 3-4
3.5
massign, ctrlinit
massign ctrlinit ichnl, insnum ichnl, ictlno1, ival1[, ictlno2, ival2 [, ictlno3, ival3[,..ival32]]
DESCRIPTION
Initialize MIDI controllers for a Csound orchestra.
ichnl MIDI channel number insnum Csound orchestra instrument number ictlno1, ictlno2, etc. MIDI contoller numbers ival1, ival2, etc. initial value for corresponding MIDI contoller number
PERFORMANCE
massign assigns a MIDI channel number to a Csound instrument ctrlinit sets initial values for a set of MIDI controllers.
AUTHORS
Barry Vercoe Mike Berry MIT, Cambridge, Mass New in Csound version 3.47
Page 3-5
Page 3-6
ORCHESTRA SYNTAX: INSTRUMENT BLOCK STATEMENTS

instr, endin
instr . . . . . endin i, j, ... < body of instrument
4.1
DESCRIPTION
These statements delimit an instrument block. They must always occur in pairs. instr begin an instrument block defining instruments i, j, ... i, j, ... must be numbers, not expressions. Any positive integer is legal, and in any order, but excessively high numbers are best avoided. endin end the current instrument block. Note: There may be any number of instrument blocks in an orchestra. Instruments can be defined in any order (but they will always be both initialized and performed in ascending instrument number order). Instrument blocks cannot be nested (i.e. one block cannot contain another).
Orchestra Syntax: Instrument Block Statements
Page 4-1
Page 4-2
ORCHESTRA SYNTAX: VARIABLE INITIALIZATION

=, init, tival, divz
= = = init init tival divz divz divz iarg karg xarg iarg iarg ia, ib, isubst ka, kb, ksubst xa, xb, ksubst
5.1
ir kr ar kr ar ir ir kr ar
DESCRIPTION
= (simple assignment) Put the value of the expression iarg (karg, xarg) into the named result. This provides a means of saving an evaluated result for later use. init Put the value of the i-time expression iarg into a k- or a-rate variable, i.e., initialize the result. Note that init provides the only case of an init-time statement being permitted to write into a perf-time (k- or a-rate) result cell; the statement has no effect at perftime. tival Put the value of the instruments internal tie-in flag into the named i-rate variable. Assigns 1 if this note has been tied onto a previously held note ( see i Statement); assigns 0 if no tie actually took place. ( see also tigoto) divz Whenever b is not zero, set the result to the value a / b; when b is zero, set it to the value of subst instead.
Orchestra Syntax: Variable Initialization
Page 5-1
Page 5-2
INSTRUMENT CONTROL: INSTRUMENT INVOCATION

schedule, schedwhen
schedule schedwhen insnum, iwhen, idur [, p4, p5,] ktrigger, kinst, kwhen, kdur [, p4, p5,]
6.1
DESCRIPTION
Adds a new score event
insnum instrument number. Equivalent to p1 in a score i statement. iwhen start time of the new event. Equivalent to p2 in a score i statement. idur duration of event. Equivalent to p3 in a score i statement.
PERFORMANCE
ktrigger trigger value for new event schedule adds a new score event. The arguments, including options, are the same as in a score. The iwhen time (p2) is measured from the time of this event. If the duration is zero or negative the new event is of MIDI type, and inherits the release sub-event from the scheduling instruction. In the case of schedwhen, the event is only scheduled when the k-rate value ktrigger is first non-zero.
E X A M PL E
;; Double hit and 1sec separation instr 1 2, 1, 0.5, p4, p5 schedule a1 p4, 60, 0.999, 0, 100, 0 shaker out a1 endin a1 instr 2 marimba out endin instr 3 table schedwhen endin p4, cpspch(p5), p6, p7, 2, 6.0, 0.05, 1, 0.1 a1
kr
kr, 1 kr, 1, 0.25, 1, p4, p5
Instrument Control: Instrument Invocation
Page 6-1
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK November, 1998 (New in Csound version 3.491) Based on work by Gabriel Maldonado
Page 6-2
6.2
schedkwhen
schedkwhen ktrigger, kmintim, kmaxnum, kinsnum, kwhen, kdur[, kp4, kp5, ]
DESCRIPTION
Adds a new score event generated by a k-rate trigger.
PERFORMANCE
ktrigger triggers a new score event. If ktrigger = 0, no new event is triggered. kmintim minimum time between generated events, in seconds. If kmintim <= 0, no time limit exists. If the kinsnum is negative (to turn off an instrument), this test is bypassed. kmaxnum maximum number of simultaneous instances of instrument kinsnum allowed. If the number of extant instances of kinsnum is >= kmaxnum, no new event is generated. If kmaxnum is <= 0, it is not used to limit event generation. If the kinsnum is negative (to turn off an instrument), this test is bypassed. kinsnum instrument number. Equivalent to p1 in a score i statement. kwhen start time of the new event. Equivalent to p2 in a score i statement. Measured from the time of the triggering event. kwhen must be >= 0. If kwhen > 0, the instrument will not be initialized until the actual time when it should start performing. kdur duration of event. Equivalent to p3 in a score i statement. If kdur = 0, the instrument will only do an initialization pass, with no performance. If kdur is negative, a held note is initiated. (See ihold and i statement.) kp4, kp5, etc. Equivalent to p4, p5, etc., in a score i statement. Note: While waiting for events to be triggered by schedkwhen, the performance must be kept going, or Csound may quit if no score events are expected. To guarantee continued performance, an f0 statement may be used in the score.
AUTHOR
Rasmus Ekman EMS Stockholm, Sweden New in Csound version 3.59
Page 6-3
6.3
turnon
turnon insnum[,itime]
DESCRIPTION
Activate an instrument, for an indefinite time.
insnum instrument number to be activated itime delay, in seconds, after which instrument insnum will be activated. Default is 0.
PERFORMANCE
turnon activates instrument insnum after a delay of itime seconds, or immediately if itime is not specified. Instrument is active until explicitly turned off. (See turnoff.)
Page 6-4
INSTRUMENT CONTROL: DURATION CONTROL STATEMENTS

ihold, turnoff
ihold turnoff
7.1
DESCRIPTION
These statements permit the current note to modify its own duration. ihold this i-time statement causes a finite-duration note to become a held note. It thus has the same effect as a negative p3 ( see score i Statement), except that p3 here remains positive and the instrument reclassifies itself to being held indefinitely. The note can be turned off explicitly with turnoff, or its space taken over by another note of the same instrument number (i.e. it is tied into that note). Effective at i-time only; no-op during a reinit pass. turnoff this p-time statement enables an instrument to turn itself off. Whether of finite duration or held, the note currently being performed by this instrument is immediately removed from the active note list. No other notes are affected.
E X A M PL E
The following statements will cause a note to terminate when a control signal passes a certain threshold (here the Nyquist frequency).
k1 if contin: a1 expon k1 < sr/2 sr turnoff oscil 440, p3/10,880 ; begin gliss and continue ; until Nyquist detected kgoto contin ; then quit a1, k1, 1
Instrument Control: Duration Control Statements
Page 7-1
Instrument Control: Duration Control Statements
Page 7-2
INSTRUMENT CONTROL: REAL-TIME PERFORMANCE CONTROL

active
active insnum
8.1
ir
DESCRIPTION
Returns the number of active instances of an instrument.
insnum number of the instrument to be reported
PERFORMANCE
active returns the number of active instances of instrument number insnum at the time it is called.
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK July, 1999 New in Csound version 3.57
Instrument Control: Real-time Performance Control
Page 8-1
8.2
cpuprc, maxalloc, prealloc

cpuprc maxalloc prealloc insnum, ipercent insnum, icount insnum, icount
DESCRIPTION
Control allocation of cpu resources on a per-instrument basis, to optimize real-time output.
insnum instrument number ipercent percent of cpu processing-time to assign. Can also be expressed as a fractional value. icount number of instrument allocations
PERFORMANCE
cpuprc sets the cpu processing-time percent usage of an instrument, in order to avoid buffer underrun in real-time performances, enabling a sort of polyphony theshold. The user must set ipercent value for each instrument to be activated in real-time. Assuming that the total theoretical processing time of the cpu of the computer is 100%, this percent value can only be defined empirically, because there are too many factors that contribute to limiting real-time polyphony in different computers. For example, if ipercent is set to 5% for instrument 1, the maximum number of voices that can be allocated in real-time, is 20 (5% * 20 = 100%). If the user attempts to play a further note while the 20 previous notes are still playing, Csound inhibits the allocation of that note and will display the following warning message: cant allocate last note because it exceeds 100% of cpu time In order to avoid audio buffer underruns, it is suggested to set the maximum number of voices slightly lower than the real processing power of the computer. Sometimes an instrument can require more processing time than normal. If, for example, the instrument contains an oscillator which reads a table that doesnt fit in cache memory, it will be slower than normal. In addition, any program running concurrently in multitasking, can subtract processing power to varying degrees. At the start, all instruments are set to a default value of ipercent = 0.0% (i.e. zero processing time or rather infinite cpu processing-speed). This setting is OK for deferredtime sessions. maxalloc limits the number of allocations of an instrument. prealloc creates space for instruments but does not run them. All instances of cpuprc, maxalloc, and prealloc must be defined in the header section, not in the instrument body.
Page 8-2
E X A M PL E
sr = kr = ksmps = nchnls = cpuprc 44100 441 100 2 1, 2.5
cpuprc 2, 33.333 instr 1 ...body... endin instr 2 ....body... endin
; set instr 1 to 2.5% of processor power, ; i.e. maximum 40 voices (2.5% * 40 = 100%) ; set instr 2 to 33.333% of processor power, ; i.e. maximum 3 voices (33.333% * 3 = 100%)
AUTHOR
Gabriel Maldonado Italy July, 1999 New in Csound version 3.57
Page 8-3
Page 8-4
INSTRUMENT CONTROL: TIME READING

timek, times, timeinstk, timeinsts
timek timek times times timeinstk timeinsts
9.1
ir kr ir kr kr kr
DESCRIPTION
Opcodes to read absolute time since the start of the performance or of an instance of an instrument in two formats.
PERFORMANCE
timek is for time in krate cycles. So with: sr = 44100 kr = 6300 ksmps = 7 then after half a second, the timek opcode would report 3150. It will always report an integer. Time in seconds is available with times. This would return 0.5 after half a second. times and timek can also operate only at the start of the instance of the instrument. Both produce an i-rate variable (starting with i or gi) as their output. timek and times can both produce a k-rate variable for output. There are no input parameters. timeinstk and timeinsts are similar to timek and times, except they return the time since the start of this instance of the instrument.
AUTHOR
Robin Whittle Australia May 1997
Instrument Control: Time Reading
Page 9-1
Page 9-2
10
INSTRUMENT CONTROL: CLOCK CONTROL

clockon, clockoff, readclock
clockon clockoff readclock inum inum inum
10.1
ir
DESCRIPTION
Starts and stops one of a number of internal clocks, and reads the value of a clock.
inum the number of a clock. There are 32 clocks numbered 0 through 31. All other values are mapped to clock number 32. ir value at i-time, of the clock specified by inum
PERFORMANCE
Between a clockon and a clockoff, the CPU time used, is accumulated in the clock. The precision is machine dependent, but is the millisecond range on UNIX and Windows systems. readclock reads the current value of a clock at initialization time. Note: there is no way to zero a clock.
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK July, 1999 New in Csound version 3.56
Instrument Control: Clock Control
Page 10-1
Page 10-2
11
INSTRUMENT CONTROL: SENSING CONTROL

pitch
pitch
AND
11.1
koct, kamp
asig, iupdte, ilo, ihi, idbthresh[, ifrqs, iconf,\\ istrt, iocts, iq, inptls, irolloff, iskip]
DESCRIPTION
Using the same techniques as spectrum and specptrk, pitch tracks the pitch of the signal in octave point decimal form, and amplitude in dB.
iupdte length of period, in seconds, that outputs are updated ilo, ihi range in which pitch is detected, expressed in octave point decimal idbthresh amplitude, expressed in decibels, necessary for the pitch to be detected. Once started it continues until it is 6 dB down. ifrqs number of divisions of an octave. Default is 12 and is limited to 120. iconf the number of conformations needed for an octave jump. Default is 10. istrt starting pitch for tracker. Default value is (ilo + ihi)/2. iocts number of octave decimations in spectrum. Default is 6. iq Q of analysis filters. Default is 10. inptls number of harmonics, used in matching. Computation time increases with the number of harmonics. Default is 4. irolloff amplitude rolloff for the set of filters expressed as fraction per octave. Values must be positive. Default is 0.6. iskip if non-zero, skips initialization
PERFORMANCE
pitch analyzes the input signal, asig, to give a pitch/amplitude pair of outputs, for the strongest frequency in the signal. The value is updated every iupdte seconds. The number of partials and rolloff fraction can effect the pitch tracking, so some experimentation may be necessary. Suggested values are 4 or 5 harmonics, with rolloff 0.6, up to 10 or 12 harmonics with rolloff 0.75 for complex timbres, with a weak fundamental.
Instrument Control: Sensing and Control
Page 11-1
AUTHOR
John ffitch University of Bath, Codemist Ltd. Bath, UK April, 1999 New in Csound version 3.54
Page 11-2
11.2
kcps, krms
pitchamdf
pitchamdf asig, imincps, imaxcps [, icps[, imedi[, idowns[,// iexcps]]]]
DESCRIPTION
Follows the pitch of a signal based on the AMDF method (Average Magnitude Difference Function). Outputs pitch and amplitude tracking signals. The method is quite fast and should run in real-time. This technique usually works best for monophonic signals.
imincps estimated minimum frequency (expressed in Hz) present in the signal imaxcps estimated maximum frequency present in the signal icps estimated initial frequency of the signal. If 0, icps = (imincps+imaxcps) / 2. The default is 0. imedi size of median filter applied to the output kcps. The size of the filter will be imedi*2+1. If 0, no median filtering will be applied. The default is 1. idowns downsampling factor for asig. Must be an integer. A factor of idowns>1 results in faster performance, but may result in worse pitch detection. Useful range is 1 4. The default is 1. iexcps how frequently pitch analysis is executed, expressed in Hz. If 0, iexcps is set to imincps. This is usually reasonable, but experimentation with other values may lead to better results. Default is 0.
PERFORMANCE
kcps pitch tracking output krms amplitude tracking output pitchamdf usually works best for monophonic signals, and is quite reliable if appropriate initial values are chosen. Setting imincps and imaxcps as narrow as possible to the range of the signals pitch, results in better detedtion and performance. Because this process can only detect pitch after an initial delay, setting icps close to the signals real initial pitch prevents spurious data at the beginning. The median filter prevents kcps from jumping. Experiment to determine the optimum value for imedi for a given signal. Other initial values can usually be left at the default settings. Lowpass filtering of asig before passing it to pitchamdf, can improve preformance, especially with complex waveforms.
Page 11-3
E X A M PL E
ginput giwave asig ftgen ftgen instr 1 loscil 1, 0, 0, -1, "input.wav", 0, 4, 0 ; input signal 2, 0, 1024, 10, 1, 1, 1, 1 ; synth wave ; get input signal ; with original freq asig, 1000 ; lowpass-filter asig, 150, 500, 200 ; extract pitch ; and envelope krms, kcps, iwave ; "resynthesize" ; with some waveform asig1 1, 1, ginput, 1
asig tone kcps, krms pitchamdf asig1 oscil out endin
AUTHOR
Peter Neubcker Munich, Germany August, 1999 New in Csound version 3.59
Page 11-4
11.3
ktemp
tempest
tempest kin, iprd, imindur, imemdur, ihp, ithresh, ihtim, ixfdbak, istartempo, ifn[, idisprd, itweek]
DESCRIPTION
Estimate the tempo of beat patterns in a control signal.
iprd period between analyses (in seconds). Typically about .02 seconds. imindur minimum duration (in seconds) to serve as a unit of tempo. Typically about .2 seconds. imemdur duration (in seconds) of the kin short-term memory buffer which will be scanned for periodic patterns. Typically about 3 seconds. ihp half-power point (in Hz) of a low-pass filter used to smooth input kin prior to other processing. This will tend to suppress activity that moves much faster. Typically 2 Hz. ithresh- loudness threshold by which the low-passed kin is center-clipped before being placed in the short-term buffer as tempo-relevant data. Typically at the noise floor of the incoming data. ihtim half-time (in seconds) of an internal forward-masking filter that masks new kin data in the presence of recent, louder data. Typically about .005 seconds. ixfdbak proportion of this units anticipated value to be mixed with the incoming kin prior to all processing. Typically about .3. istartempo initial tempo (in beats per minute). Typically 60. ifn table number of a stored function (drawn left-to-right) by which the short-term memory data is attenuated over time. idisprd (optional) if non-zero, display the short-term past and future buffers every idisprd seconds (normally a multiple of iprd). The default value is 0 (no display). itweek (optional) fine-tune adjust this unit so that it is stable when analyzing events controlled by its own output. The default value is 1 (no change).
PERFORMANCE
tempest examines kin for amplitude periodicity, and estimates a current tempo. The input is first low-pass filtered, then center-clipped, and the residue placed in a short-term memory buffer (attenuated over time) where it is analyzed for periodicity using a form of autocorrelation. The period, expressed as a tempo in beats per minute, is output as ktemp. The period is also used internally to make predictions about future amplitude patterns, and these are placed in a buffer adjacent to that of the input. The two adjacent buffers can be periodically displayed, and the predicted values optionally mixed with the incoming signal to simulate expectation. This unit is useful for sensing the metric implications of any k-signal (e.g.- the RMS of an audio signal, or the second derivative of a conducting gesture), before sending to a tempo statement.
Page 11-5
E X A M PL E
ksum ktemp specsum tempest wsignal, 1 ; sum the amps of a spectrum ksum, .02, .1, 3, 2, 800, .005, 0, 60, 4, .1, .995 ; and look for beats
Page 11-6
11.4
ar
follow
follow asig, idt
DESCRIPTION
Envelope follower unit generator.
idt This is the period, in seconds, that the average amplitude of asig is reported. If the frequency of asig is low then idt must be large (more than half the period of asig )
PERFORMANCE
asig This is the signal from which to extract the envelope.
E X A M PL E
k1 a1 ak1 a2 0, p3, 30000 ; Make k1 a simple envelope line k1, 1000, 1 ; Make a simple signal using k1 oscil ; ak1 is now like k1 follow a1, .02 ak1, 1000, 1 ; Make a simple signal using ak1 oscil a2 ; Both a1 and a2 are the same out To avoid zipper noise, by discontinuities produced from complex envelope tracking, a lowpass filter could be used, to smooth the estimated envelope.
AUTHOR
Paris Smaragdis MIT, Cambridge 1995
Page 11-7
11.5
kout
trigger
trigger ksig, kthreshold, kmode
DESCRIPTION
Informs when a krate signal crosses a threshold.
PERFORMANCE
ksig input signal kthreshold trigger threshold kmode can be 0 , 1 or 2 Normally trigger outputs zeroes: only each time ksig crosses kthreshold trigger outputs a 1. There are three modes of using ktrig: kmode = 0 (down-up) ktrig outputs a 1 when current value of ksig is higher than kthreshold, while old value of ksig was equal to or lower than kthreshold. kmode = 1 (up-down) ktrig outputs a 1 when current value of ksig is lower than kthreshold while old value of ksig was equal or higher than kthreshold. kmode = 2 (both) ktrig outputs a 1 in both the two previous cases.
AUTHOR
Gabriel Maldonado Italy New in Csound version 3.49
Page 11-8
11.6
kr kr
peak
peak peak ksig asig
DESCRIPTION
These opcodes maintain the output k-rate variable as the peak absolute level so far received.
PERFORMANCE
kr Output equal to the highest absolute value received so far. This is effectively an input to the opcode as well, since it reads kr in order to decide whether to write something higher into it. ksig k-rate input signal. asig a-rate input signal.
DEPRECATED NAME
Prior to Csound version 3.63, the k-rate version of peak was called peakk. peak is now used with either k- or a-rate input.
AUTHOR
Page 11-9
11.7
kx, ky
xyin, tempo
xyin tempo iprd, ixmin, ixmax, iymin, iymax[, ixinit, iyinit] ktempo, istartempo
DESCRIPTION
Sense the cursor position in an output window. Apply tempo control to an uninterpreted score. When xyin is called the position of the mouse within the output window is used to reply to the request. This simple mechanism does mean that only one xyin can be used accurately at once. The position of the mouse is reported in the output window.
iprd- period of cursor sensing (in seconds). Typically .1 seconds. xmin, xmax, ymin, ymax edge values for the x-y coordinates of a cursor in the input window. ixinit, iyinit (optional) initial x-y coordinates reported; the default values are 0,0. If these values are not within the given min-max range, they will be coerced into that range. istartempo initial tempo (in beats per minute). Typically 60.
PERFORMANCE
xyin samples the cursor x-y position in an input window every iprd seconds. Output values are repeated (not interpolated) at the k-rate, and remain fixed until a new change is registered in the window. There may be any number of input windows. This unit is useful for real-time control, but continuous motion should be avoided if iprd is unusually small. tempo allows the performance speed of Csound scored events to be controlled from within an orchestra. It operates only in the presence of the Csound -t flag. When that flag is set, scored events will be performed from their uninterpreted p2 and p3 (beat) parameters, initially at the given command-line tempo. When a tempo statement is activated in any instrument (ktempo 0.), the operating tempo will be adjusted to ktempo beats per minute. There may be any number of tempo statements in an orchestra, but coincident activation is best avoided.
E X A M PL E
kx,ky xyin tempo .05, 30, 0, 120, 0, 75 ; sample the cursor kx, 75 ; and control the tempo of performance
Page 11-10
11.8
ar
follow2
follow2 asig, katt, krel
DESCRIPTION
A controllable envelope extractor using the algorithm attributed to Jean-Marc Jot.
PERFORMANCE
asig the input signal whose envelope is followed katt the attack rate (60dB attack time in seconds) krel the decay rate (60dB decay time in seconds) The output tracks the amplitude envelope of the input signal. The rate at which the output grows to follow the signal is controlled by the katt, and the rate at which it decreases in response to a lower amplitude, is controlled by the krel. This gives a smoother envelope than follow.
E X A M PL E
a1 follow2 ain, 0.01, .1
AUTHOR
John ffitch University of Bath, Codemist Ltd. Bath, UK February, 2000 New in Csound version 4.03
Page 11-11
11.9
kout
setctrl, control
setctrl control inum, kval, itype knum
DESCRIPTION
Configurable slider controls for realtime user input. Requires Winsound or TCL/TK. setctrl sets a slider to a specific value, or sets a minimum or maximum range. control reads a slider's value.
inum number of the slider to set itype type of value sent to the slider as follows: 1 set the current value. Initial value is 0. 2 set the minimum value. Default is 0. 3 set the maximum value. Default is 127. 4 set the label. (New in Csound version 4.09)
PERFORMANCE
kval value to be sent to the slider Calling setctrl or control will create a new slider on the screen. There is no theoretical limit to the number of sliders. Windows and TCL/TK use only integers for slider values, so the values may need rescaling. GUIs usually pass values at a fairly slow rate, so it may be advisable to pass the output of control through port.
E X A M PL E
#define SLIDERNUM # 6 # instr 1 kgoto continue ; We don't want to configure sliders at k; rate! 20, 0 10, 1 1000, 2 ; Read values with smoothing
; Set min=10, max=1000, actual=20 $SLIDERNUM., setctrl $SLIDERNUM., setctrl $SLIDERNUM., setctrl continue: kcHz $SLIDERNUM. control kcHz kcHz, .02 port ; ... etc endin
AUTHOR
John ffitch University of Bath, Codemist. Ltd. Bath, UK July, 2000 New in Csound version 4.06
Page 11-12
11.10 b u t t o n , c h e c k b o x
kr kr button checkbox inum inum
DESCRIPTION
Sense on-screen controls. Needs Windows or TCL/TK.
inum the number of the button or checkbox. If it does not exist, it is made on-screen at initialization.
PERFORMANCE
If the button has been pushed since the last k-period, then return 1, otherwise return 0. If the checkbox is set (pushed) then return 1, if not, return 0.
E X A M PL E
Increase pitch while a checkbox is set, and extend duration for each push of a button.
kcps k1 a1 k2 instr 1 = cpsoct(p5) 1 check if (k1 == 1) kcps = kcps * 1.1 p4, kcps, 1 oscil a1 out button 1 if (k2 == 1) p3 = p3 + 0.1 endin
AUTHOR
John ffitch University of Bath, Codemist Ltd. Bath, UK September, 2000 New in Csound version 4.08
Page 11-13
11.11 s e n s e k e y
kr sensekey
DESCRIPTION
Returns the ASCII code of a key that has been pressed, or -1 if no key has been pressed.
PERFORMANCE
At release, this has not been properly verified, and seems not to work at all on Windows.
AUTHOR
John ffitch University of Bath, Codemist. Ltd. Bath, UK October, 2000 New in Csound version 4.09
Page 11-14
12
INSTRUMENT CONTROL: CONDITIONAL VALUES

>, <, >=, <=, ==, !=, ?
(a (a (a (a (a (a > < >= <= == != b b b b b b ? ? ? ? ? ? v1 v1 v1 v1 v1 v1 : : : : : : v2) v2) v2) v2) v2) v2)
12.1
DESCRIPTION
where a, b, v1 and v2 may be expressions, but a, b not audio-rate. In the above conditionals, a and b are first compared. If the indicated relation is true (a greater than b, a less than b, a greater than or equal to b, a less than or equal to b, a equal to b, a not equal to b), then the conditional expression has the value of v1; if the relation is false, the expression has the value of v2. (For convenience, a sole = will function as = =.) NB.: If v1 or v2 are expressions, these will be evaluated before the conditional is determined. In terms of binding strength, all conditional operators (i.e. the relational operators (<, etc.), and ?, and : ) are weaker than the arithmetic and logical operators (+, -, *, /, && and ||). These are operators not opcodes. Therefore, they can be used within orchestra statements, but do not form complete statements themselves.
E X A M PL E
k2 = (k1 < p5/2 + p6 ? k1 : p7)
binds the terms p5/2 and p6. It will return the value k1 below this threshold, else the value p7.
Instrument Control: Conditional Values
Page 12-1
Page 12-2
13
13.1
INSTRUMENT CONTROL: MACROS

#define, $NAME, #undef
#define #define $NAME. #undef NAME # replacement text # NAME(a b c) # replacement text # NAME
DESCRIPTION
Macros are textual replacements which are made in the orchestra as it is being read. The macro system in Csound is a very simple one, and uses the characters # and $ to define and call macros. This can save typing, and can lead to a coherent structure and consistent style. This is similar to, but independent of, the macro system in the score language. #define NAME defines a simple macro. The name of the macro must begin with a letter and can consist of any combination of letters and numbers. Case is significant. This form is limiting, in that the variable names are fixed. More flexibility can be obtained by using a macro with arguments, described below. #define NAME(a b c) defines a macro with arguments. This can be used in more complex situations. The name of the macro must begin with a letter and can consist of any combination of letters and numbers. Within the replacement text, the arguments can be substituted by the form: $A. In fact, the implementation defines the arguments as simple macros. There may be up to 5 arguments, and the names may be any choice of letters. Remember that case is significant in macro names. $NAME. calls a defined macro. To use a macro, the name is used following a $ character. The name is terminated by the first character which is neither a letter nor a number. If it is necessary for the name not to terminate with a space, a period, which will be ignored, can be used to terminate the name. The string, $NAME., is replaced by the replacement text from the definition. The replacement text can also include macro calls. #undef NAME undefines a macro name. If a macro is no longer required, it can be undefined with #undef NAME.
# replacement text # The replacement text is any character string (not containing a #) and can extend over multiple lines. The replacement text is enclosed within the # characters, which ensure that additional characters are not inadvertently captured.
PERFORMANCE
Some care is needed with textual replacement macros, as they can sometimes do strange things. They take no notice of any meaning, so spaces are significant. This is why, unlike the C programming language, the definition has the replacement text surrounded by # characters. Used carefully, this simple macro system is a powerful concept, but it can be abused.
Instrument Control: Macros
Page 13-1
E X A M PL E S
Simple Macro
#define REVERB #ga = ga+a1 out a1# a1 $REVERB. instr 1 oscil endin instr 2 repluck endin endin
a1 $REVERB.
This will get expanded before compilation into:

a1 ga instr 1 oscil = out a1 endin instr 2 repluck = out endin
ga+a1
a1 ga
ga+a1 a1
Macro With Arguments

#define REVERB(A) #ga = ga+$A. out $A.# instr 1 a1 oscil $REVERB(a1) endin instr 2 a2 repluck $REVERB(a2) endin
This will get expanded before compilation into:

a1 ga instr 1 oscil = out a1 endin instr 2 repluck = out a2 endin
ga+a1
a2 ga
ga+a2
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK April, 1998 (New in Csound version 3.48)
Page 13-2
13.2
#include
#include filename
DESCRIPTION:
It is sometimes convenient to have the orchestra arranged in a number of files, for example with each instrument in a separate file. This style is supported by the #include facility which is part of the macro system. A line containing the text #include filename where the character can be replaced by any suitable character. For most uses the double quote symbol will probably be the most convenient. The file name can include a full path. This takes input from the named file until it ends, when input reverts to the previous input. There is currently a limit of 20 on the depth of included files and macros. Another suggested use of #include would be to define a set of macros which are part of the composers style. An extreme form would be to have each instrument defines as a macro, with the instrument number as a parameter. Then an entire orchestra could be constructed from a number of #include statements followed by macro calls.
#include clarinet #include flute #include bassoon $CLARINET(1) $FLUTE(2) $BASSOON(3)
It must be stressed that these changes are at the textual level and so take no cognizance of any meaning.
AUTHOR
Page 13-3
Page 13-4
14
INSTRUMENT CONTROL: PROGRAM FLOW CONTROL

igoto, tigoto, kgoto, goto, if, timout
igoto tigoto kgoto goto if if if timout label label label label ia R ib igoto label ka R kb kgoto label ia R ib goto label istrt, idur, label
14.1
DESCRIPTION
where label is in the same instrument block and is not an expression, and where R is one of the Relational operators (<, =, <=, ==, !=) (and = for convenience, see also under Conditional Values). These statements are used to control the order in which statements in an instrument block are to be executed. i-time and p-time passes can be controlled separately as follows: igoto During the i-time pass only, unconditionally transfer control to the statement labeled by label. tigoto similar to igoto, but effective only during an i-time pass at which a new note is being tied onto a previously held note ( see i Statement); no-op when a tie has not taken place. Allows an instrument to skip initialization of units according to whether a proposed tie was in fact successful (see also tival, delay). kgoto During the p-time passes only, unconditionally transfer control to the statement labeled by label. goto (combination of igoto and kgoto) Transfer control to label on every pass. if...igoto conditional branch at i-time, depending on the truth value of the logical expression ia R ib. The branch is taken only if the result is true. if...kgoto conditional branch during p-time, depending on the truth value of the logical expression ka R kb. The branch is taken only if the result is true. if...goto combination of the above. Condition tested on every pass. timout conditional branch during p-time, depending on elapsed note time. istrt and idur specify time in seconds. The branch to label will become effective at time istrt, and will remain so for just idur seconds. Note that timout can be reinitialized for multiple activation within a single note ( see example under reinit).
E X A M PL E
if k3 p5 + 10 kgoto next
Instrument Control: Program Flow Control
Page 14-1
Page 14-2
15
INSTRUMENT CONTROL: REINITIALIZATION

reinit, rigoto, rireturn
reinit rigoto rireturn label label
15.1
DESCRIPTION
These statements permit an instrument to reinitialize itself during performance. reinit whenever this statement is encountered during a p-time pass, performance is temporarily suspended while a special Initialization pass, beginning at label and continuing to rireturn or endin, is executed. Performance will then be resumed from where it left off. rigoto similar to igoto, but effective only during a reinit pass (i.e., no-op at standard itime). This statement is useful for bypassing units that are not to be reinitialized. rireturn terminates a reinit pass (i.e., no-op at standard i-time). This statement, or an endin, will cause normal performance to be resumed.
E X A M PL E
The following statements will generate an exponential control signal whose value moves from 440 to 880 exactly ten times over the duration p3.
reset: contin: timout reinit expon rireturn 0, p3 /10, contin ; after p3/10 seconds, reset ; reinit both timout 440, p3/10,880 ; and expon ; then resume perf
Instrument Control: Reinitialization
Page 15-1
Page 15-2
16
MATHEMATICAL OPERATIONS: ARITHMETIC AND LOGIC OPERATIONS

-, +, &&, ||, *, /, ^, %
+ && || + * / ^ % a a b b b b b b b b (no rate restriction) (no rate restriction) (logical AND; not audio-rate) (logical OR; not audio-rate) (no rate restriction) (no rate restriction) (no rate restriction) (no rate restriction) (b not audio-rate) (no rate restriction)
16.1
a a a a a a a a
DESCRIPTION
where the arguments a and b may be further expressions. Arithmetic operators perform operations of change-sign (negate), dont-change-sign, logical AND logical OR, add, subtract, multiply and divide. Note that a value or an expression may fall between two of these operators, either of which could take it as its left or right argument, as in
a + b * c.
In such cases three rules apply: 1. * and / bind to their neighbors more strongly than + and -. Thus the above expression is taken as
a + (b * c)
with * taking b and c and then + taking a and b * c. 2. + and bind more strongly than &&, which in turn is stronger than ||:
a && b c || d
is taken as
(a && (b c)) || d
3. When both operators bind equally strongly, the operations are done left to right:
a b c
is taken as
(a b) c
Parentheses may be used as above to force particular groupings.
The Public Csound Reference Manual Version 4.10Mathematical Operations: Arithmetic and Logic Operations Page 16-1
The operator ^ raises a to the b power. b may not be audio-rate. Use with caution as precedence may not work correctly. See Section 5.2. New in Csound version 3.493. The operator % returns the value of a reduced by b, so that the result, in absolute value, is that of the absolute value of b, by repeated subtraction. This is the same as modulus function in integers. New in Csound version 3.50.
The Public Csound Reference Manual Version 4.10Mathematical Operations: Arithmetic and Logic Operations Page 16-2
17
MATHEMATICAL OPERATIONS: MATHEMATICAL FUNCTIONS

int, frac, i, abs, exp, log, log10, sqrt
int(x) int frac(x) frac i(x) abs(x) abs exp(x) exp log(x) log log10(x) log10 sqrt(x) sqrt (init- or control-rate args only) (init- or control-rate args only) (control-rate args only) (no rate restriction) (no rate restriction) (no rate restriction) (no rate restriction) (no rate restriction)
17.1
DESCRIPTION
Where the argument within the parentheses may be an expression. These functions perform arithmetic translation from units of one kind to units of another. The result can then be a term in a further expression. int(x) returns the integer part of x. frac(x) returns the fractional part of x. i(x) returns an init-type equivalent of the argument (k-rate) abs(x) returns the absolute value of x. exp(x) returns e raised to the xth power. log(x) returns the natural log of x (x positive only). log10(x) returns the base 10 log of x (x positive only). sqrt(x) returns the square root of x (x non-negative). Note that for log, log10, and sqrt the argument value is restricted.
Mathematical Operations: Mathematical Functions
Page 17-1
17.2
powoftwo, logbtwo
powoftwo(x) (init-rate or control-rate args only) powoftwo logbtwo(x) (init-rate or control-rate args only) logbtwo
DESCRIPTION
Power-of-two operations.
PERFORMANCE
powoftwo() function returns 2 ^ x and allows positive and negatives numbers as argument. The range of values admitted in powoftwo() is -5 to +5 allowing a precision more fine than one cent in a range of ten octaves. If a greater range of values is required, use the slower opcode pow. logbtwo() returns the logarithm base two of x. The range of values admitted as argument is .25 to 4 (i.e. from -2 octave to +2 octave response). This function is the inverse of powoftwo(). These functions are fast, because they read values stored in tables. Also they are very useful when working with tuning ratios. They work at i- and k-rate.
AUTHORS
Gabriel Maldonado Italy June, 1998 John ffitch University of Bath, Codemist, Ltd. Bath, UK July, 1999 New in Csound version 3.57
Page 17-2
18
MATHEMATICAL OPERATIONS: TRIGONOMETRIC FUNCTIONS

sin, cos, tan, sininv, cosinv, taninv, sinh, cosh, tanh
sin(x) sin cos(x) cos tan(x) tan sininv(x) sininv cosinv(x) cosinv taninv(x) taninv sinh(x) sinh cosh(x) cosh tanh(x) tanh (no (no (no (no (no (no (no (no (no rate rate rate rate rate rate rate rate rate restriction) restriction) restriction) restriction) restriction) restriction) restriction) restriction) restriction)
18.1
DESCRIPTION
Where the argument within the parentheses may be an expression. These functions perform trigonometric conversions. The result can then be a term in a further expression. sin(x) returns the sine of x (x in radians). cos(x) returns the cosine of x (x in radians). tan (x) returns the tangent of x. sininv(x) returns the arcsine of x. cosinv(x) returns the arcosine of x. taninv(x) returns the arctangent of x. sinh(x) returns the hyperbolic sine of x. cosh(x) returns the hyperbolic cosine of x. tanh (x) returns the hyperbolic tangent of x .
Mathematical Operations: Trigonometric Functions
Page 18-1
Page 18-2
19
MATHEMATICAL OPERATIONS: AMPLITUDE FUNCTIONS

dbamp, ampdb dbfsamp, ampdbfs
dbamp(x) dbamp ampdb(x) ampdb dbfsamp(x) dbfsamp ampdbfs(x) ampdbfs (init- or control-rate args only) (no rate restriction) (init- or control-rate args only) (no rate restriction)
19.1
DESCRIPTION
Where the argument within the parentheses may be an expression. These functions perform conversions between raw amplitude values and their decibel equivelents. The result can then be a term in a further expression. dbamp(x) returns the decibel equivalent of the raw amplitude x. ampdb(x) returns the amplitude equivalent of the decibel value x. Thus: 60 dB = 1000 66 dB = 1995.262 72 dB = 3891.07 78 dB = 7943.279 84 dB = 15848.926 90 dB = 31622.764 dbfsamp(x) returns the decibel equivalent, relative to full scale amplitude, of the raw amplitude x. Full scale is assumed to be 16 bit. New is Csound version 4.10. ampdbfs(x) returns the amplitude equivalent of the decibel value x, which is relative to full scale amplitude. Full scale is assumed to be 16 bit. New is Csound version 4.10.
Mathematical Operations: Amplitude Functions
Page 19-1
Page 19-2
20
MATHEMATICAL OPERATIONS: RANDOM FUNCTIONS

rnd, birnd
rnd(x) rnd birnd(x) birnd (init- or control-rate args only) (init- or control-rate args only)
20.1
DESCRIPTION
Where the argument within the parentheses may be an expression. These value converters sample a global random sequence, but do not reference seed. The result can be a term in a further expression.
PERFORMANCE
rnd(x) returns a random number in the unipolar range 0 to x. birnd(x) returns a random number in the bipolar range -x to x. rnd and birnd obtain values from a global pseudo-random number generator, then scale them into the requested range. The single global generator will thus distribute its sequence to these units throughout the performance, in whatever order the requests arrive
AUTHOR
Barry Vercoe MIT Cambridge, Massachusetts 1997
Mathematical Operations: Random Functions
Page 20-1
Page 20-2
21
MATHEMATICAL FUNCTIONS: OPCODE EQUIVALENTS OF FUNCTIONS

sum
sum asig1, asig2[, asig3...asigN]
21.1
ar
DESCRIPTION
Sums any number of a-rate signals.
PERFORMANCE
asig1, etc. a-rate signals to be summed (mixed or added).
AUTHOR
Gabriel Maldonado Italy April, 1999 New in Csound version 3.54
The Public Csound Reference Manual Version 4.10Mathematical Functions: Opcode Equivalents of Functions
Page 21-1
21.2
ar
product
product asig1, asig2[, asig3...asigN]
DESCRIPTION
Multiplies any number of a-rate signals.
PERFORMANCE
asig1, etc. a-rate signals to be multiplied.
AUTHOR
Gabriel Maldonado Italy April, 1999 New in Csound version 3.54
Page 21-2
21.3
ir ir kr ar
pow
pow = pow pow iarg, ipow iarg ^ ipow karg, kpow[, inorm] aarg, kpow[, inorm]
DESCRIPTION
Computes xarg to the power of kpow (or ipow) and scales the result by inorm.
inorm The number to divide the result (default to 1). This is especially useful if you are doing powers of a- or k- signals where samples out of range are extremely common! iarg i-rate base ipow i-rate exponent
PERFORMANCE
karg k-rate base. kpow k-rate exponent aarg a-rate base.
E X A M PL E S
i2t2 kline kexp pow line pow 2,2 0, 1, 4 kline, 2, 4 ; Computes 2^2.
This feeds a linear function to pow and scales that to the lines peak value. The output will be an exponential curve with the same range as the input line.
iamp a1 a2 pow oscil pow out 10, 2 iamp, 100, 1 a1, 2, iamp a2
This will output a sine with its negative part folded over the amplitude axis. The peak value will be iamp = 10^2 = 100. The first line could also be written:
i2t2 = 2 ^ 2
Use ^ with caution in arithmetical statements, as the precedence may not be correct. This operator is new as of Csound version 3.493.
DEPRECATED NAMES
pow was originally three opcodes called ipow, kpow, and apow. As of Csound version 3.48 those names are deprecated, and the three seperate opcodes replaced by pow.
AUTHOR
Page 21-3
21.4
ir kr ar
taninv2
taninv2 taninv2 taninv2 ix, iy kx, ky ax, ay
DESCRIPTION
Returns the arctangent of iy/ix, ky/kx, or ay/ax. If either x or y is zero, taninv2 returns zero.
ix, iy values to be converted
PERFORMANCE
kx, ky control rate signals to be converted ax, ay audio rate signals to be converted
AUTHOR
Page 21-4
21.5
ar ar
mac, maca
mac maca asig1, ksig1, asig2[, ksig2, asig3, ...asigN, ksigN] asig1, asig2[, asig3, asig4, asig5, ...asigN]
DESCRIPTION
Multiply and accumulate k- and/or a-rate signals.
PERFORMANCE
ksig1, etc. k-rate input signals asig1, etc. a-rate input signals mac multiplies and accumulates a- and k-rate signals. It is equivalent to:
ar = asig1 + ksig1*asig2 + ksig2+asig3 + ...
maca multiplies and accumulates a-rate signals only. It is equivalent to:

ar = asig1 + asig2*asig3 + asig4+asig5 + ...
AUTHOR
John ffitch University of Bath, Codemist, Ltd. Bath, UK May, 1999 New in Csound version 3.55
Page 21-5
Page 21-6
22
22.1
PITCH CONVERTERS: FUNCTIONS

octpch, pchoct, cpspch, octcps, cpsoct
octpch pchoct cpspch octcps cpsoct (pch) (oct) (pch) (cps) (oct) (init- or control-rate (init- or control-rate (init- or control-rate (init- or control-rate (no rate restriction) args args args args only) only) only) only)
DESCRIPTION
where the argument within the parentheses may be a further expression. These are really value converters with a special function of manipulating pitch data. Data concerning pitch and frequency can exist in any of the following forms: Name Abbreviation
octave point pitch-class (8ve.pc) octave point decimal cycles per second
pch oct cps (Hz)
The first two forms consist of a whole number, representing octave registration, followed by a specially interpreted fractional part. For pch, the fraction is read as two decimal digits representing the 12 equal-tempered pitch classes from .00 for C to.11 for B. For oct, the fraction is interpreted as a true decimal fractional part of an octave. The two fractional forms are thus related by the factor 100/12. In both forms, the fraction is preceded by a whole number octave index such that 8.00 represents Middle C, 9.00 the C above, etc. Thus A440 can be represented alternatively by 440 (cps),8.09 (pch), 8.75 (oct), or 7.21 (pch), etc. Microtonal divisions of the pch semitone can be encoded by using more than two decimal places. The mnemonics of the pitch conversion units are derived from morphemes of the forms involved, the second morpheme describing the source and the first morpheme the object (result). Thus
cpspch(8.09) cpspch
will convert the pitch argument 8.09 to its cps (or Hertz) equivalent, giving the value of 440. Since the argument is constant over the duration of the note, this conversion will take place at i-time, before any samples for the current note are produced. By contrast, the conversion cpsoct(8.75 + k1) cpsoct which gives the value of A440 transposed by the octave interval k1 will repeat the calculation every, k-period since that is the rate at which k1 varies. Note: The conversion from pch or oct into cps is not a linear operation but involves an exponential process that could be time-consuming when executed repeatedly. Csound now uses a built-in table lookup to do this efficiently, even at audio rates.
Pitch Converters: Functions
Page 22-1
Page 22-2
23
PITCH CONVERTERS: TUNING OPCODES

cps2pch, cpsxpch
cps2pch cpsxpch ipch, iequal ipch, iequal, irepeat, ibase
23.1
icps icps
DESCRIPTION
Converts a pitch-class notation into cycles-per-second (Hz) for equal divisions of the octave (for cps2pch) or for equal divisions of any interval. There is a restriction of no more than 100 equal divisions.
ipch Input number of the form 8ve.pc, indicating an octave and which note in the octave. iequal if positive, the number of equal intervals into which the octave is divided. Must be less than or equal to 100. If negative, is the number of a table of frequency multipliers. irepeat Number indicating the interval which is the octave. The integer 2 corresponds to octave divisions, 3 to a twelfth, 4 is two octaves, and so on. This need not be an integer, but must be positive. ibase The frequency which corresponds to pitch 0.0 Note: The following are essentially the same
ia ib ic = cps2pch cpsxpch cpspch(8.02) cpspch 8.02, 12 8.02, 12, 2, 1.02197503906
These are opcodes not functions. Negative values of ipch are allowed, but not negative irepeat, iequal or ibase.
E X A M PL E
; convert oct.pch to ; cps in 19ET inote p5, 12, 3, 261.62561 ; Pierce scale centered cpsxpch ; on middle A inote p5, 21, 4, 16.35160062496 ; 10.5ET scale cpsxpch The use of a table allows exotic scales by mapping frequencies in a table. For example the table: f2 0 16 -2 1 1.1 1.2 1.3 1.4 1.6 1.7 1.8 1.9 can be used with: ip p4, -2 cps2pch to get a 10 note scale of unequal divisions. inote cps2pch p5, 19
Pitch Converters: Tuning Opcodes
Page 23-1
AUTHOR
John ffitch University of Bath/Codemist Ltd. Gabriel Maldonado Italy 1998 (New in Csound version 3.492) Bath, UK 1997
Pitch Converters: Tuning Opcodes
Page 23-2
24
24.1
MIDI S U P P O R T : C O N V E R T E R S
notnum, veloc, cpsmidi, cpsmidib, octmidi, octmidib, pchmidi, pchmidib, ampmidi, aftouch, pchbend, midictrl
notnum veloc cpsmidi cpsmidib cpsmidib octmidi octmidib octmidib pchmidi pchmidib pchmidib ampmidi aftouch pchbend pchbend midictrl midictrl [ilow, ihigh] [irange] [irange] [irange] [irange] [irange] [irange] iscal[, ifn] [imin[, imax]] [imin[, imax]] [imin[, imax]] inum[imin[, imax]] inum[imin[, imax]]
ival ival icps icps kcps ioct ioct koct ipch ipch kpch iamp kaft ibend kbend ival kval
DESCRIPTION
Get a value from the MIDI event that activated this instrument, or from a continuous MIDI controller, and convert it to a locally useful format.
iscal i-time scaling factor. ifn (optional) function table number of a normalized translation table, by which the incoming value is first interpreted. The default value is 0, denoting no translation. inum, ictlno MIDI controller number initial the initial value of the controller ilow, ihigh low and high ranges for mapping irange the pitch bend range in semitones ichnl the MIDI channel imin, imax set minimum and maximum limits on values obtained
MIDI Support: Converters
Page 24-1
PERFORMANCE
notnum, veloc get the MIDI byte value (0 127) denoting the note number or velocity of the current event. cpsmidi, octmidi, pchmidi get the note number of the current MIDI event, expressed in cps, oct, or pch units for local processing. cpsmidib, octmidib, pchmidib get the note number of the current MIDI event, modify it by the current pitch-bend value, and express the result in cps, oct, or pch units. Available as an i-time value or as a continuous k-rate value. ampmidi get the velocity of the current MIDI event, optionally pass it through a normalized translation table, and return an amplitude value in the range 0 iscal. aftouch, pchbend get the current after-touch, or pitch-bend value for this channel, rescaled to the range 0 iscal. Note that this access to pitch-bend data is independent of the MIDI pitch, enabling the value here to be used for any arbitrary purpose. midictrl get the current value (0 127) of a specified MIDI controller.
AUTHOR
Barry Vercoe Mike Berry MIT Mills May 1997
Page 24-2
24.2
icps
cpstmid
cpstmid ifn
DESCRIPTION
This unit is similar to cpsmidi, but allows fully customized micro-tuning scales.
ifn function table containing the parameters (numgrades, interval, basefreq, basekeymidi) and the tuning ratios.
PERFORMANCE
Init-rate only cpsmid requires five parameters. The first, ifn, is the function table number of the tuning ratios, and the other parameters must be stored in the function table itself. The function table ifn should be generated by GEN2, with normalization inhibited. The first four values stored in this function are: 1. numgrades the number of grades of the micro-tuning scale 2. interval the frequency range covered before repeating the grade ratios,for example 2 for one octave, 1.5 for a fifth etc. 3. basefreq the base frequency of the scale in Hz 4. basekeymidi the MIDI note number to which basefreq is assigned unmodified After these four values, the user can begin to insert the tuning ratios. For example, for a standard 12 note scale with the base frequency of 261 Hz assigned to the key number 60, the corresponding f statement in the score to generate the table should be:
; numgrades basefreq ; interval f1 0 64 -2 12 2 261 1.189207115003 ..etc... tuning-ratios (equal temp) basekeymidi 60 1 1.059463094359 1.122462048309
Another example with a 24 note scale with a base frequency of 440 assigned to the key number 48, and a repetition interval of 1.5:
; numgrades basefreq ; interval basekeymidi f1 0 64 -2 24 1.5 440 48 tuning-ratios ....... 1 1.01 1.02 1.03 ..etc...
AUTHOR
Gabriel Maldonado Italy 1998 (New in Csound version 3.492)
Page 24-3
Page 24-4
25
25.1
MIDI S U P P O R T : C O N T R O L L E R I N P U T
initc7, initc14, initc21
initc7 initc14 initc21 ichan, ictlno, ivalue ichan, ictlno1, ictlno2, ivalue ichan, ictlno1, ictlno2, ictlno3, ivalue
DESCRIPTION
Initializes MIDI controller ictlno with ivalue
ichan MIDI channel ictlno controller number (initc7) ictlno1 most significant byte controller number ictlno2 in initc14 least significant byte controller number; in initc21 Medium Significant Byte controller number ictlno3 least significant byte controller number ivalue floating point value (must be within 0 to 1)
PERFORMANCE
initc7, initc14, initc21 can be used together with both midicXX and ctrlXX opcodes for initializing the first controllers value. ivalue argument must be set with a number within 0 to 1. An error occurs if it is not. Use the following formula to set ivalue according with midicXX and ctrlXX min and max range:
ivalue = (initial_value min) / (max min)
AUTHOR
MIDI Support: Controller Input
Page 25-1
25.2
midic7, midic14, midic21, ctrl7, ctrl14, ctrl21

midic7 midic7 midic14 midic14 midic21 midic21 ctrl7 ctrl7 ctrl14 ctrl14 ctrl21 ctrl21 ictlno, imin, imax [, ifn] ictlno, kmin, kmax [, ifn] ictlno1, ictlno2, imin, imax [, ifn] ictlno1, ictlno2, kmin, kmax [, ifn] ictlno1, ictlno2, ictlno3, imin, imax [, ifn] ictlno1, ictlno2, ictlno3, kmin, kmax [, ifn] ichan, ictlno, imin, imax [,ifn] ichan, ictlno, kmin, kmax [,ifn] ichan, ictlno1, ictlno2, imin, imax [,ifn] ichan, ictlno1, ictlno2, kmin, kmax [,ifn] ichan, ictlno1, ictlno2, ictlno3, imin, imax [,ifn] ichan, ictlno1, ictlno2, ictlno3, kmin, kmax [,ifn]
idest kdest idest kdest idest kdest idest kdest idest kdest idest kdest
DESCRIPTION
Allow precise MIDI input controller signal.
idest output signal ictlno MIDI controller number (1-127) ictlno1 most-significant byte controller number (1-127) ictlno2 in midic14: least-significant byte controller number (1-127); in midic21: midsignificant byte controller number (1-127) ictlno3 least-significant byte controller number (1-127) imin user-defined minimum floating-point value of output imax user-defined maximum floating-point value of output ifn (optional) table to be read when indexing is required. Table must be normalized. Output is scaled according to imax and imin val.
PERFORMANCE
kdest output signal kmin user-defined minimum floating-point value of output kmax user-defined maximum floating-point value of output midic7 (i- and k-rate 7 bit MIDI control) allows floating point 7 bit MIDI signal scaled with a minimum and a maximum range. It also allows optional non-interpolated table indexing. In midic7 minimum and maximum values can be varied at k-rate. midic14 (i- and k-rate 14 bit MIDI control) do the same as the above with 14 bit precision. midic21 (i- and k-rate 21 bit MIDI control) do the same as the above with 21 bit precision. midic14 and midic21 can use optional interpolated table indexing. They require two or three MIDI controllers as input.
Page 25-2
ctrl7, ctrl14, ctrl21 are very similar to midicXX opcodes the only differences are: ctrlXX UGs can be included in score oriented instruments without Csound crashes. They need the additional parameter ichan containing the MIDI channel of the controller. MIDI channel is the same for all the controller used in a single ctrl14 or ctrl21 opcode.
NAMES
DEPRECATED
The opcode names imidic7, imidic14, imidic21, ictrl7, ictrl14, and ictrl21 have been deprecated in Csound version 3.52. Instead use imidic7, imidic14, imidic21, ictrl7, ictrl14, and ictrl21, respectively, with i-rate outputs.
AUTHOR
Page 25-3
25.3
ival kval
chanctrl
chanctrl chanctrl ichnl, ictlno[,ilow,ihigh] ichnl, ictlno[,ilow,ihigh]
DESCRIPTION
Get the current value of a controller and optionally map it onto specified range.
ichnl the MIDI channel ictlno the MIDI controller number ilow, ihigh low and high ranges for mapping
AUTHOR
Mike Berry Mills College May 1997
Page 25-4
26
26.1
MIDI S U P P O R T : S L I D E R B A N K S
slider8, slider16, slider32, slider64, slider8f, slider16f, slider32f, slider64f, s16b14, s32b14
slider8 ichan, ictlnum1, imin1, imax1, ifn1, ....,\\ ictlnum8, imin8, imax8, ifn8 ichan, ictlnum1, imin1, imax1, init1, ifn1,\\ slider8 ..., ictlnum8, imin8, imax8, init8, ifn8 ichan, ictlnum1, imin1, imax1, ifn1, ....,\\ slider16 ictlnum16, imin16, imax16, ifn16 ichan, ictlnum1, imin1, imax1, init1, ifn1,\\ slider16 ...., ictlnum16, imin16, imax16, init16, ifn16 ichan, ictlnum1, imin1, imax1, ifn1, ....,\\ slider32 ictlnum32, imin32, imax32, ifn32 slider32 ichan, ictlnum1, imin1, imax1, init1, fn1,\\ ...., ictlnum32, imin32, imax32, init32, ifn32 ichan, ictlnum1, imin1, imax1, ifn1, ....,\\ slider64 ictlnum64, imin64, imax64, ifn64 ichan, ictlnum1, imin1, imax1, init1, ifn1,\\ slider64 ...., ictlnum64, imin64, imax64, init64, ifn64 ichan, ictlnum1, imin1, imax1, init1, ifn1,\\ slider8f icutoff1, ...., ictlnum8, imin8, imax8, init8,\\ ifn8, icutoff8 slider16f ichan, ictlnum1, imin1, imax1, init1, ifn1,\\ icutoff1, .... ,ictlnum16, imin16, imax16,\\ init16, ifn16, icutoff16 slider32f ichan, ictlnum1, imin1, imax1, init1, ifn1,\\ icutoff1, .... , ictlnum32, imin32, imax32,\\ init32, ifn32, icutoff32 slider64f ichan, ictlnum1, imin1, imax1, init1, ifn1,\\ icutoff1, .... , ictlnum64, imin64, imax64,\\ init64, ifn64, icutoff64 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1,\\ s16b14 initvalue1, ifn1, ....., ictlno_msb16,\\ ictlno_lsb16, imin16, imax16, initvalue16, ifn16 s16b14 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1,\\ ifn1, ictlno_msb16, ictlno_lsb16, imin16,\\ imax16, ifn16 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1,\\ s32b14 initvalue1, ifn1, ....., ictlno_msb32,\\ ictlno_lsb32, imin32, imax32, initvalue32, ifn32 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1,\\ s32b14 ifn1, ...., ictlno_msb32, ictlno_lsb32, imin32,\\ imax32, ifn32
i1, ..., i8 k1, ..., k8 i1, ..., i16 k1, ..., k16 i1, ..., i32 k1, ..., k32 i1, ..., i64 k1, ..., k64 k1, ..., k8 k1, ..., k16 k1, ..., k32 k1, ..., k64 i1, ..., i16 k1, ..., k16 i1, ..., i32 k1, ..., k32
DESCRIPTION
MIDI slider control banks
MIDI Support: Slider Banks
Page 26-1
i1 ... i64 output values ichan MIDI channel (1-16) ictlnum1 ... ictlnum64 MIDI control number ictlno_msb1 .... ictlno_msb32 MIDI control number (most significant byte) ictlno_lsb1 .... ictlno_lsb32 MIDI control number (least significant byte) imin1 ... imin64 minimum values for each controller imax1 ... imax64 maximum values for each controller init1 ... init64 initial value for each controller ifn1 ... ifn64 function table for conversion for each controller icutoff1 ... icutoff64 low-pass filter cutoff frequency for each controller
PERFORMANCE
k1 ... k64 output values sliderN and sliderNf are banks of MIDI controllers, useful when using MIDI mixer such as Kawai MM-16 or others for changing whatever sound parameter in real-time. The raw MIDI control messages at the input port are converted to agree with iminN and imaxN, and an initial value can be set. Also, an optional non-interpolated function table with a custom translation curve is allowed, useful for enabling exponential response curves. When no function table translation is required, set the ifnN value to 0, else set ifnN to a valid function table number. When table translation is enabled (i.e. setting ifnN value to a non-zero number referring to an already allocated function table), initN value should be set equal to iminN or imaxN value, else the initial output value will not be the same as specified in initN argument. slider8 allows a bank of 8 different MIDI control message numbers, slider16 does the same with a bank of 16 controls, and so on. sliderNf filter the signal before output, for eliminating discontinuities due to the low resolution of the MIDI (7 bit); the cutoff frequency can be set separately for each controller (suggested range: .1 to 5 Hz). As the input and output arguments are many, you can split the line using \ (backslash) character (new in 3.47 version) to improve the readability. Using these opcodes is considerably more efficient than using the separate ones (ctrl7 and tonek) when more controllers are required. In the i-rate version of sliderN, there is not an initial value input argument, because the output is gotten directly from current status of internal controller array of Csound. sNb14 opcode is the 14-bit version of this bank of controllers. Warning: sliderNf opcodes do not output the required initial value immediately, but only after some k-cycles, because the filter slightly delays the output.
Page 26-2
DEPRECATED NAMES
The opcode names islider8, islider16, islider32, islider64, is16b14, and is32b14 have been deprecated as of Csound version 3.52. Use slider8, slider16, slider32, slider64, s16b14, and s32b14, respectively, for i-rate output.
AUTHOR
Gabriel Maldonado Italy December 1998 (New in Csound version 3.50)
Page 26-3
Page 26-4
27
27.1
MIDI S U P P O R T : G E N E R I C I/O
midiin
midiin
kstatus, kchan, kdata1, kdata2
DESCRIPTION
Returns a generic MIDI message received by the MIDI IN port
PERFORMANCE
kstatus the type of MIDI message. Can be: 128 (note off), 144 (note on), 160 (polyphonic aftertouch), 176 (control change), 192 (program change), 208 (channel aftertouch), 224 (pitch bend) 0 if no MIDI messages are pending in the MIDI IN buffer.
kchan MIDI channel (1-16) kdata1, kdata2 message-dependent data values midiin has no input arguments, because it reads at the MIDI in port implicitly. It works at krate. Normally (i.e., when no messages are pending) kstatus is zero, only when MIDI data are present in the MID IN buffer, is kstatus set to the type of the relevant messages.
AUTHOR
MIDI Support: Generic I/O
Page 27-1
27.2
midiout
midiout kstatus, kchan, kdata1, kdata2
DESCRIPTION
Sends a generic MIDI message to the MIDI OUT port
PERFORMANCE
kstatus the type of MIDI message. Can be: 128 (note off), 144 (note on), 160 (polyphonic aftertouch), 176 (control change), 192 (program change), 208 (channel aftertouch), 224 (pitch bend) 0 when no MIDI messages must be sent to the MIDI OUT port.
kchan MIDI channel (1-16) kdata1, kdata2 message-dependent data values midiout has no output arguments, because it sends a message to the MIDI OUT port implicitly. It works at k-rate. It sends a MIDI message only when kstatus is non-zero. Warning: Normally kstatus should be set to 0. Only when the user intends to send a MIDI message, can it be set to the corresponding message type number.
AUTHOR
Page 27-2
28
28.1
MIDI S U P P O R T : N O T E - O N /N O T E - O F F
noteon, noteoff, noteondur, noteondur2
noteon noteoff noteondur noteondur2 ichn, ichn, ichn, ichn, inum, inum, inum, inum, ivel ivel ivel, idur ivel, idur
DESCRIPTION
Send note-on and note-off messages to the MIDI OUT port.
ichn MIDI channel number (0-15) inum note number (0-127) ivel velocity (0-127)
PERFORMANCE
noteon (i-rate note on) and noteoff (i-rate note off) are the simplest MIDI OUT opcodes. noteon sends a MIDI noteon message to MIDI OUT port, and noteoff sends a noteoff message. A noteon opcode must always be followed by an noteoff with the same channel and number inside the same instrument, otherwise the note will play endlessly. These noteon and noteoff are useful only when introducing a timout statement to play a nonzero duration MIDI note. For most purposes it is better to use noteondur and noteondur2. noteondur and noteondur2 (i-rate note on with duration) send a noteon and a noteoff MIDI message both with the same channel, number and velocity. Noteoff message is sent after idur seconds are elapsed by the time noteondur was active. noteondur differs from noteondur2 in that noteondur truncates note duration when current instrument is deactivated by score or by real-time playing, while noteondur2 will extend performance time of current instrument until idur seconds have elapsed. In realtime playing it is suggested to use noteondur also for undefined durations, giving a large idur value. Any number of noteondur or noteondur2 opcodes can appear in the same Csound instrument, allowing chords to be played by a single instrument.
N A M E C HA N G ES
Prior to Csound version 3.52 (February, 1999), these opcodes were called ion, ioff, iondur, and iodur2. ondur and ondur2 changed to noteondur and noteondur2 in Csound version 3.53.
MIDI Support: Note-on/Note-off
Page 28-1
AUTHOR
Page 28-2
28.2
moscil, midion
moscil midion kchn, knum, kvel, kdur, kpause kchn, knum, kvel
DESCRIPTION
Send a stream of note-on and note-off messages to the MIDI OUT port.
PERFORMANCE
kchn MIDI channel number (0-15) knum note number (0-127) kvel velocity (0-127) kdur note duration in seconds kpause pause duration after each noteoff and before new note in seconds moscil and midion are the most powerful MIDI OUT opcodes. moscil (MIDI oscil) plays a stream of notes of kdur duration. Channel, pitch, velocity, duration and pause can be controlled at k-rate, allowing very complex algorithmically generated melodic lines. When current instrument is deactivated, the note played by current instance of moscil is forcedly truncated. midion (k-rate note on) plays MIDI notes with current kchn, knum and kvel. These arguments can be varied at k-rate. Each time the MIDI converted value of any of these arguments changes, last MIDI note played by current instance of midion is immediately turned off and a new note with the new argument values is activated. This opcode, as well as moscil, can generate very complex melodic textures if controlled by complex k-rate signals. Any number of moscil or midion opcodes can appear in the same Csound instrument, allowing a counterpoint-style polyphony within a single instrument.
DEPRECATED NAMES
midion was originally called kon. As of Csound version 3.493, that name is deprecated. midion should be used instead of kon.
AUTHOR
Gabriel Maldonado Italy May 1997 (moscil new in Csound version 3.47)
Page 28-3
28.3
midion2
midion2 kchn, knum, kvel, ktrig
DESCRIPTION
Sends noteon and noteoff messages to the MIDI out port when triggered by a value different than zero.
PERFORMANCE
kchn MIDI channel knum MIDI note number kvel note velocity ktrig trigger input signal (normally 0) Similar to midion, this opcode sends noteon and noteoff messages to the MIDI out port, but only when ktrig is non-zero. This opcode is can work together with the output of the trigger opcode.
AUTHOR
Page 28-4
29
MIDI S U P P O R T : MIDI M E S S A G E OUTPUT

outic, outkc, outic14, outkc14, outipb, outkpb, outiat, outkat, outipc, outkpc, outipat, outkpat
ichn, kchn, ichn, kchn, ichn, kchn, ichn, kchn, ichn, kchn, inum, knum, imsb, kmsb, ivalue, imin, kvalue, kmin, ilsb, ivalue, klsb, kvalue, imax kmax imin, imax kmin, kmax
29.1
outic outkc outic14 outkc14 outipb outkpb outiat outkat outipc outkpc outipat outkpat
ivalue, imin, imax kvalue, kmin, kmax ivalue, imin, imax kvalue, kmin, kmax iprog, imin, imax kprog, kmin, kmax
ichn, inotenum, ivalue, imin, imax kchn, knotenum, kvalue, kmin, kmax
DESCRIPTION
Send a single Channel message to the MIDI OUT port.
PERFORMANCE
ichn, kchn MIDI channel number (0-15) inum, knum controller number (0-127 for example 1 = ModWheel; 2 = BreathControl etc.) ivalue, kvalue floating point value imin, kmin minimum floating point value (converted in MIDI integer value 0) imax, kmax maximum floating point value (converted in MIDI integer value 127 (7 bit) or 16383 (14 bit)) imsb, kmsb most significant byte controller number when using 14 bit parameters ilsb, klsb least significant byte controller number when using 14 bit parameters iprog, kprog program change number in floating point inotenum, knotenum MIDI note number (used in polyphonic aftertouch messages) outic and outkc (i- and k-rate MIDI controller output) send controller messages to MIDI OUT device. outic14 and outkc14 (i and k-rate MIDI 14 bit controller output) send a pair of controller messages. These opcodes can drive 14 bit parameters on MIDI instruments that recognize them. The first control message contains the most significant byte of i(k)value argument while the second message contains the less significant byte. i(k)msb and i(k)lsb are the number of the most and less significant controller.
MIDI Support: MIDI Message Output
Page 29-1
outipb and outkpb (i- and k-rate pitch bend output) send pitch bend messages. outiat and outkat (i- and k-rate aftertouch output) send aftertouch messages. outiat and outkat (i- and k-rate aftertouch output) send aftertouch messages. outipc and outkpc (i- and k-rate program change output) send program change messages. outipat and outkpat (i- and k-rate polyphonic aftertouch output) send polyphonic aftertouch messages. These opcodes can drive a different value of a parameter for each note currently active. They work only with MIDI instruments which recognize them. N.B. All these opcodes can scale the i(k)value floating-point argument according with i(k)max and i(k)min values. For example, setting i(k)min = 1.0 and i(k)max = 2.0, when i(k)value argument receives a 2.0 value, the opcode will send a 127 value to MIDI OUT device, while when receiving a 1.0 it will send a 0 value. i-rate opcodes send their message once during instrument initialization. k-rate opcodes send a message each time the MIDI converted value of argument i(k)value changes.
DEPRECATED NAMES
Prior to Csound version 3.52, these opcodes were named ioutc, koutc, ioutc14, koutc14, ioutpb, koutpb, ioutat, koutat, ioutpc, koutpc, ioutpat, and koutpat. The current names were adopted with version 3.52 (February, 1999) to avoid name space pollution.
AUTHOR
Page 29-2
29.2
nrpn
nrpn kchan, kparmnum, kparmvalue
DESCRIPTION
Sends a NPRN (Non Registered Parameter Number) message to the MIDI OUT port each time one of the input arguments changes.
PERFORMANCE
kchan MIDI channel kparmnum number of NRPN parameter kparmvalue value of NRPN parameter This opcode sends new message when the MIDI translated value of one of the input arguments changes. It operates at k-rate. Useful with the MIDI instruments that recognize NRPNs (for example with the newest sound-cards with internal MIDI synthesizer such as SB AWE32, AWE64, GUS etc. in which each patch parameter can be changed during the performance via NRPN)
AUTHOR
Page 29-3
29.3
mdelay
mdelay kstatus, kchan, kd1, kd2, kdelay
DESCRIPTION
A MIDI delay opcode.
PERFORMANCE
kstatus status byte of MIDI message to be delayed kchan MIDI channel (1-16) kd1 first MIDI data byte kd2 second MIDI data byte kdelay delay time in seconds Each time that kstatus is other than zero, mdelay outputs a MIDI message to the MIDI out port after kdelay seconds. This opcode is useful in implementing MIDI delays. Several instances of mdelay can be present in the same instrument with different argument values, so complex and colorful MIDI echoes can be implemented. Further, the delay time can be changed at k-rate.
AUTHOR
Gabriel Maldonado Italy November, 1998 (New in Csound version 3.492)
Page 29-4
30
MIDI S U P P O R T : R E A L - T I M E MESSAGES
mclock, mrtmsg
mclock mrtmsg ifreq imsgtype
30.1
DESCRIPTION
Send system real-time messages to the MIDI OUT port.
ifreq clock message frequency rate in Hz imsgtype type of real-time message: 1 sends a START message (0xFA) 2 sends a CONTINUE message (0xFB) 0 sends a STOP message (0xFC) -1 sends a SYSTEM RESET message (0xFF) -2 sends an ACTIVE SENSING message (0xFE)
PERFORMANCE
mclock (MIDI clock) sends a MIDI CLOCK message (0xF8) every 1/ifreq seconds. So ifreq is the frequency rate of CLOCK message in Hz. mrtmsg (MIDI real-time message) sends a real-time message once, in init stage of current instrument. imsgtype parameter is a flag to indicate the message type (see above).
AUTHOR
MIDI Support: Real-time Messages
Page 30-1
MIDI Support: Real-time Messages
Page 30-2
31
31.1
kflag
MIDI S U P P O R T : E V E N T E X T E N D E R S
xtratim, release
xtratim release iextradur
DESCRIPTION
Extend the duration of real-time generated events and handle their extra life (see also linenr).
iextradur additional duration of current instrument instance
PERFORMANCE
xtratim extends current MIDI-activated note duration of iextradur seconds after the corresponding note-off message has deactivated current note itself. This opcode has no output arguments. release outputs current note state. If current note is in the release stage (i.e. if its duration has been extended with xtratim opcode and if it has only just deactivated), kflag output argument is set to 1, else (in sustain stage of current note) is set to 0. These two opcodes are useful for implementing complex release-oriented envelopes.
E X A M PL E
instr 1 ;allows complex ADSR envelope with MIDI events inum notnum icps cpsmidi iamp 4000 ampmidi ; ;------- complex envelope block -----1 ;extra-time, i.e. release dur xtratim krel 0 init krel ;outputs release-stage flag (0 or 1 values) release ;if in release-stage goto release section if (krel .5) kgoto rel ; ;************ attack and sustain section *********** kmp1 0, .03, 1, .05, 1, .07, 0, .08, .5, 4, 1, 50, 1 linseg kmp = kmp1*iamp kgoto done ; ;--------- release section -------rel: kmp2 1, .3, .2, .7, 0 linseg kmp = kmp1*kmp2*iamp done: ;-----a1 kmp, icps, 1 oscili a1 out endin
MIDI Support: Event Extenders
Page 31-1
AUTHOR
MIDI Support: Event Extenders
Page 31-2
32
SIGNAL GENERATORS: LINEAR EXPONENTIAL GENERATORS
AND
32.1
line, expon, linseg, linsegr, expseg, expsegr, expsega

line line expon expon linseg linseg linsegr linsegr expseg expseg expsegr expsegr expsega ia, ia, ia, ia, ia, ia, ia, ia, ia, ia, ia, ia, ia, idur1, idur1, idur1, idur1, idur1, idur1, idur1, idur1, idur1, idur1, idur1, idur1, idur1, ib ib ib ib ib[, ib[, ib[, ib[, ib[, ib[, ib[, ib[, ib[,
kr ar kr ar kr ar kr ar kr ar kr ar ar
idur2, idur2, idur2, idur2, idur2, idur2, idur2, idur2, idur2,
ic[...]] icI...]] ic[...]], icI...]], ic[...]] ic[...]] ic[...]], ic[...]], ic[...]]
irel, iz irel, iz irel, iz irel, iz
DESCRIPTION
Output values kr or ar trace a straight line (exponential curve) or a series of line segments (or exponential segments) between specified points.
ia- starting value. Zero is illegal for exponentials. ib, ic, etc. value after dur1 seconds, etc. For exponentials, must be non-zero and must agree in sign with ia. idur1 duration in seconds of first segment. A zero or negative value will cause all initialization to be skipped. idur2, idur3, etc. duration in seconds of subsequent segments. A zero or negative value will terminate the initialization process with the preceding point, permitting the lastdefined line or curve to be continued indefinitely in performance. The default is zero. irel, iz duration in seconds and final value of a note releasing segment.
PERFORMANCE
These units generate control or audio signals whose values can pass through 2 or more specified points. The sum of dur values may or may not equal the instruments performance time: a shorter performance will truncate the specified pattern, while a longer one will cause the last-defined segment to continue on in the same direction. linsegr, expsegr are amongst the Csound r units that contain a note-off sensor and release time extender. When each senses an event termination or MIDI noteoff, it immediately extends the performance time of the current instrument by irel seconds, and sets out to reach the value iz by the end of that period (no matter which segment the unit is in). r units can also be modified by MIDI noteoff velocities (see veloffs). For two or more extenders in an instrument, extension is by the greatest period.
The Public Csound Reference Manual Version 4.10 Signal Generators: Linear and Exponential Generators Page 32-1
expsega is almost identical to expseg, but more precise when defining segments with very short durations (i.e., in a percussive attack phase) at audio rate. Note that expseg does not operate correctly at audio rate when segments are shorter than a k-period. In this situation, expsega should be used instead of expseg.
E X A M PL E
k2 expseg 440, p3/2,880, p3/2,440
This statement creates a control signal which moves exponentially from 440 to 880 and back, over the duration p3.
AUTHOR
Gabriel Maldonado (expsega) Italy June, 1998 New in Csound version 3.57
Signal Generators: Linear and Exponential Generators
Page 32-2
32.2
kr kr kr kr ar ar ar ar
adsr, madsr, xadsr, mxadsr

adsr madsr xadsr mxadsr adsr madsr xadsr mxadsr iatt, iatt, iatt, iatt, iatt, iatt, iatt, iatt, idec, idec, idec, idec, idec, idec, idec, idec, islev, islev, islev, islev, islev, islev, islev, islev, irel[, irel[, irel[, irel[, irel[, irel[, irel[, irel[, idel] idel] idel] idel] idel] idel] idel] idel]
DESCRIPTION
Calculates the classical ADSR envelope
iatt duration of attack phase idec duration of decay islev level for sustain phase irel duration of release phase idel period of zero before the envelope starts
PERFORMANCE
The envelope is the range 0 to 1 and may need to be scaled further. The envelope may be described as:
The length of the sustain is calculated from the length of the note. This means adsr is not suitable for use with MIDI events. The opcode madsr uses the linsegr mechanism, and so can be used in MIDI applications. The opcodes xadsr and mxadsr are identical to adsr and madsr, respectively, except they use exponential, rather than linear, line segments. adsr and madsr new in Csound version 3.49. xadsr and mxadsr new in Csound version 3.51.
Page 32-3
32.3
kr ar
transeg
transeg transeg ibeg, idur, itype, ival ibeg, idur, itype, ival
DESCRIPTION
Constructs a user-definable envelope.
I N i T I A L I ZA T I O N
ibeg starting value ival value after dur seconds idur duration in seconds of segment itype if 0, a straight line is produced. If non-zero, then transeg creates the following curve, for n steps: ibeg+(ivalue-ibeg)*(1-exp(i*itype/(n-1)))/(1-exp(itype))
PERFORMANCE
If itype > 0, there is a slowly rising, fast decaying (convex) curve, while if itype < 0, the curve is fast rising, slowly decaying (concave). See also GEN16.
AUTHOR
Page 32-4
33
33.1
SIGNAL GENERATORS: TABLE ACCESS

table, tablei, table3, oscil1, oscil1i, osciln
table tablei table3 table tablei table3 table tablei table3 oscil1 oscil1i osciln indx, indx, indx, kndx, kndx, kndx, andx, andx, andx, idel, idel, kamp, ifn[, ifn[, ifn[, ifn[, ifn[, ifn[, ifn[, ifn[, ifn[, kamp, kamp, ifrq, ixmode[, ixoff[, ixmode[, ixoff[, ixmode[, ixoff[, ixmode[, ixoff[, ixmode[, ixoff[, ixmode[, ixoff[, ixmode[, ixoff[, ixmode[, ixoff[, ixmode[, ixoff[, idur, ifn idur, ifn ifn, itimes iwrap]]] iwrap]]] iwrap]]] iwrap]]] iwrap]]] iwrap]]] iwrap]]] iwrap]]] iwrap]]]
ir ir ir kr kr kr ar ar ar kr kr ar
DESCRIPTION
Table values are accessed by direct indexing or by incremental sampling.
ifn function table number. tablei, oscil1i require the extended guard point. ixmode (optional) index data mode. The default value is 0. 0 = raw index 1 = normalized (0 to 1) ixoff (optional) amount by which index is to be offset. For a table with origin at center, use tablesize/2 (raw) or .5 (normalized). The default value is 0. iwrap (optional) wraparound index flag. The default value is 0. 0 = nowrap (index < 0 treated as index=0; index tablesize sticks at index=size) 1 = wraparound idel delay in seconds before oscil1 incremental sampling begins. idur duration in seconds to sample through the oscil1 table just once. A zero or negative value will cause all initialization to be skipped. ifrq, itimes rate and number of times through the stored table.
PERFORMANCE
table invokes table lookup on behalf of init, control or audio indices. These indices can be raw entry numbers (0,l,2...size 1) or scaled values (0 to 1-e). Indices are first modified by the offset value then checked for range before table lookup (see iwrap). If index is likely to be full scale, or if interpolation is being used, the table should have an extended guard point. table indexed by a periodic phasor ( see phasor) will simulate an oscillator. oscil1 accesses values by sampling once through the function table at a rate determined by idur. For the first idel seconds, the point of scan will reside at the first location of the table; it will then begin moving through the table at a constant rate, reaching the end in
The Public Csound Reference Manual Version 4.10 Signal Generators: Table Access Page 33-1
another idur seconds; from that time on (i.e. after idel + idur seconds) it will remain pointing at the last location. Each value obtained from sampling is then multiplied by an amplitude factor kamp before being written into the result. Because oscil1 is an interpolating opcode, the table it reads should have a guard point. osciln will sample several times through the stored table at a rate of ifrq times per second, after which it will output zeros. Generates audio signals only, with output values scaled by kamp. tablei and oscil1i are interpolating units in which the fractional part of index is used to interpolate between adjacent table entries. The smoothness gained by interpolation is at some small cost in execution time (see also oscili, etc.), but the interpolating and noninterpolating units are otherwise interchangeable. Note that when tablei uses a periodic index whose modulo n is less than the power of 2 table length, the interpolation process requires that there be an (n+ 1)th table value that is a repeat of the 1st (see f Statement in score). table3 is experimental, and is identical to tablei, except that it uses cubic interpolation. (New in Csound version 3.50.)
Signal Generators: Table Access
Page 33-2
34
34.1
kr ar
SIGNAL GENERATORS: PHASORS

phasor
phasor phasor kcps[, iphs] xcps[, iphs]
DESCRIPTION
Produce a normalized moving phase value.
iphs (optional) initial phase, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to he skipped. The default value is zero.
PERFORMANCE
An internal phase is successively accumulated in accordance with the kcps or xcps frequency to produce a moving phase value, normalized to lie in the range 0. <= phs < 1. When used as the index to a table unit, this phase (multiplied by the desired function table length) will cause it to behave like an oscillator. Note that phasor is a special kind of integrator, accumulating phase increments that represent frequency settings.
E X A M PL E
k1 kpch a1 phasor table oscil 1 ; cycle once per second k1 * 12, 1 ; through 12-note pch table p4, cpspch(kpch), 2 ; with continuous sound
Signal Generators: Phasors
Page 34-1
34.2
kr ar
phasorbnk
phasorbnk phasorbnk kcps, kndx, icnt[, iphs] xcps, kndx, icnt[, iphs]
DESCRIPTION
Produce an arbitrary number of normalized moving phase values, accessable by an index.
34.2.1
I N I T I A L I ZA T I O N icnt maximum number of phasors to be used.

iphs initial phase, expressed as a fraction of a cycle (0 to 1). If -1 initialization is skipped. If iphas>1 each phasor will be initialized with a random value.
34.2.2
PERFORMANCE
kndx index value to access individual phasors For each independent phasor, an internal phase is successively accumulated in accordance with the kcps or xcps frequency to produce a moving phase value, normalized to lie in the range 0 <= phs < 1. Each individual phasor is accessed by index kndx. This phasor bank can be used inside a k-rate loop to generate multiple independent voices, or together with the adsynt opcode to change parameters in the tables used by adsynt.
34.2.3
E X A M PL E Generate multiple voices with independent partials. This example is better with adsynt. See also the example under adsynt, for k-rate use of phasorbnk.
giwave icnt asum kindex loop: kcps aphas asig asum ftgen instr 1 = = = = phasorbnk table = 1, 0, 1024, 10, 1 10 0 0 ; generate a sinewave table
; generate 10 voices ; empty output buffer ; reset loop index ; loop executed every k-cycle ; non-harmonic partials ; get phase for each voice ; and read wave from table ; accumulate output ; do loop
(kindex+1)*100 + 30 kcps, kindex, icnt aphas, giwave, 1 asum + asig kindex + 1 kgoto loop asum*3000
kindex = if (kindex < icnt) out endin
AUTHOR
Page 34-2
35
SIGNAL GENERATORS: BASIC OSCILLATORS

oscil, oscili, oscil3
oscil oscili oscil3 oscil oscili oscil3 kamp, kamp, kamp, xamp, xamp, xamp, kcps, kcps, kcps, xcps, xcps, xcps, ifn[, ifn[, ifn[, ifn[, ifn[, ifn[, iphs] iphs] iphs] iphs] iphs] iphs]
35.1
kr kr kr ar ar ar
DESCRIPTION
Table ifn is incrementally sampled modulo the table length and the value obtained is multiplied by amp.
ifn function table number. Requires a wrap-around guard point. iphs (optional) initial phase of sampling, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is 0.
PERFORMANCE
The oscil units output periodic control (or audio) signals consisting of the value of kamp(xamp)times the value returned from control rate (audio rate) sampling of a stored function table. The internal phase is simultaneously advanced in accordance with the kcps or xcps input value. While the amplitude and frequency inputs to the k-rate oscils are scalar only, the corresponding inputs to the audio-rate oscils may each be either scalar or vector, thus permitting amplitude and frequency modulation at either sub-audio or audio frequencies. oscili differs from oscil in that the standard procedure of using a truncated phase as a sampling index is here replaced by a process that interpolates between two successive lookups. Interpolating generators will produce a noticeably cleaner output signal, but they may take as much as twice as long to run. Adequate accuracy can also be gained without the time cost of interpolation by using large stored function tables of 2K, 4K or 8K points if the space is available. oscil3 is experimental, and is identical to oscili, except that it uses cubic interpolation. (New in Csound version 3.50.)
E X A M PL E
k1 a1 oscil oscil 10, 5, 1 5000, 440 + k1, 1 ; 5 Hz vibrato ; around A440 + -10 Hz
Signal Generators: Basic Oscillators
Page 35-1
35.2
ar kr ar kr
poscil, poscil3
poscil poscil poscil3 poscil3 kamp, kamp, kamp, kamp, kcps, kcps, kcps, kcps, ifn ifn ifn ifn [,iphs] [,iphs] [,iphs] [,iphs]
DESCRIPTION
High precision oscillators. poscil3 uses cubic interpolation.
ifn function table number iphs (optional) initial phase of sampling, expressed as a fraction of a cycle (0 to 1). The default value is 0.
PERFORMANCE
ar output signal kamp amplitude kcps frequency poscil (precise oscillator) is the same as oscili, but allows much more precise frequency control, especially when using long tables and low frequency values. It uses floating-point table indexing, instead of integer math, like oscil and oscili. It is only a bit slower than oscili.
AUTHORS
Gabriel Maldonado (poscil) Italy 1998 (New in Csound version 3.52) John ffitch (poscil3) University of Bath/Codemist Ltd. Bath, UK February, 1999 (New in Csound version 3.52)
Page 35-2
35.3
kr ar
lfo
lfo lfo kamp, kcps[, itype] kamp, kcps[, itype]
DESCRIPTION
A low frequency oscillator of various shapes.
itype -- determine the waveform of the oscillator. Default is 0. 0: 1: 2: 3: 4: 5: sine triangles square (bipolar) square (unipolar) saw-tooth saw-tooth(down)
The sine wave is implemented as a 4096 table and linear interpolation. The others are calculated.
PERFORMANCE
kamp amplitude of output kcps frequency of oscillator
E X A M PL E
kp ar instr 1 lfo oscil out endin 10, 5, 4 p4, p5+kp, 1 ar
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK November, 1998 (New in Csound version 3.491)
Page 35-3
Page 35-4
36
SIGNAL GENERATORS: DYNAMIC SPECTRUM OSCILLATORS

buzz, gbuzz
buzz gbuzz xamp, xcps, knh, ifn[, iphs] xamp, xcps, knh, klh, kr, ifn[, iphs]
36.1
ar ar
DESCRIPTION
Output is a set of harmonically related cosine partials.
ifn table number of a stored function containing (for buzz) a sine wave, or (for gbuzz) a cosine wave. In either case a large table of at least 8192 points is recommended. iphs (optional) initial phase of the fundamental frequency, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is zero
PERFORMANCE
The buzz units generate an additive set of harmonically related cosine partials of fundamental frequency xcps, and whose amplitudes are scaled so their summation peak equals xamp. The selection and strength of partials is determined by the following control parameters: knh total number of harmonics requested. New in Csound version 3.57, knh defaults to one. If knh is negative, the absolute value is used. klh lowest harmonic present. Can be positive, zero or negative. In gbuzz the set of partials can begin at any partial number and proceeds upwards; if klh is negative, all partials below zero will reflect as positive partials without phase change (since cosine is an even function), and will add constructively to any positive partials in the set. kr specifies the multiplier in the series of amplitude coefficients. This is a power series: if the klhth partial has a strength coefficient of A, the (klh + n)th partial will have a coefficient of A * (kr ** n), i.e. strength values trace an exponential curve. kr may be positive, zero or negative, and is not restricted to integers. buzz and gbuzz are useful as complex sound sources in subtractive synthesis. buzz is a special case of the more general gbuzz in which klh = kr = 1; it thus produces a set of knh equal-strength harmonic partials, beginning with the fundamental. (This is a band-limited pulse train; if the partials extend to the Nyquist, i.e. knh = int (sr / 2 / fundamental freq.), the result is a real pulse train of amplitude xamp.) Although both knh and klh may be varied during performance, their internal values are necessarily integer and may cause pops due to discontinuities in the output; kr, however, can be varied during performance to good effect. Both buzz and gbuzz can be amplitude- and/or frequency-modulated by either control or audio signals.
Signal Generators: Dynamic Spectrum Oscillators
Page 36-1
N.B. These two units have their analogs in GEN11, in which the same set of cosines can be stored in a function table for sampling by an oscillator. Although computationally more efficient, the stored pulse train has a fixed spectral content, not a time-varying one as above.
Page 36-2
36.2
ar
vco
vco kamp, kfqc[, iwave][, ipw][, ifn][, imaxd]
DESCRIPTION
Implementation of a band limited, analog modeled oscillator, based on integration of band limited impulses. vco can be used to simulate a variety of analog wave forms. Last four arguments were made optional in Csound version 4.10.
iwave (optional) determines the waveform : 1: sawtooth 2: Square/PWM 3: triangle/Saw/Ramp
iwave defaults to 1. ipw (optional) determines the pulse width when iwave is set to 2, and determines Saw/Ramp character when iwave is set to 3. The value of ipw should be between 0 and 1. A value of .5 will generate a square wave or a triangle wave depending on iwave. Default is 1. ifn (optional) the table number of a of a stored sine wave. Default is 1. imaxd (optional) is the maximum delay time. A time of 1/ifqc may be required for the pwm and triangle waveform. To bend the pitch down this value must be as large as 1/(minimum frequency). Default is 1.
PERFORMANCE
kamp determines the amplitude kfqc is the frequency of the wave
E X A M PL E
idur iamp ifqc iwave isine imaxd kpw1 kpw asig instr 10 = = = = = = oscil = vco outs endin p3 ; Duration p4 ; Amplitude cpspch(p5) cpspch p6 ; Frequency ; Selected wave form 1=Saw, ; 2=Square/PWM, 3=Tri/Saw-Ramp-Mod
1 1/ifqc*2 ; Allows pitch bend down of two octaves .25, ifqc/200, 1 kpw1 + .5 iamp, ifqc, iwave, kpw, 1, imaxd asig, asig ; Output and amplification
Page 36-3
f1 0 65536 10 1 ; Sta i10 0 i10 + i10 . i10 . i10 . i10 . i10 . i10 . i10 . i10 . i10 . i10 . e Dur 2 . . 2 . . 2 . . 2 . . Amp 20000 . . 20000 . . 20000 . . 20000 . . Pitch 5.00 . 2 . 3 7.00 . 2 . 3 9.00 . 2 . 3 11.00 . . Wave 1 1 1 1 2 3
AUTHOR
Hans Mikelson December, 1998 (New in Csound version 3.50)
Page 36-4
36.3
ar
mpulse
mpulse kamp, kfreq[, ioffset]
DESCRIPTION
Generate a set of impulses. of amplitude kamp at frequency kfreq. The first impulse is after a delay of ioffset seconds. The value of kfreq is read only after an impulse, so it is the interval to the next impulse at the time of an impulse.
ioffset the delay before the first impulse. If it is negative, the value is taken as the number of samples, otherwise it is in seconds. Default is zero.
PERFORMANCE
kamp amplitude of the impulses generated kfreq frequency of the impulse train After the initial delay, an impulse of kamp amplitude is generated as a single sample. Immediately after generating the impulse, the time of the next one is calculated. If kfreq is zero, there is an infinite wait to the next impulse. If kfreq is negative, the frequency is counted in samples rather than seconds.
E X A M PL E
Generate a set of impulses at 10 a second, after a delay of 0.05s
a1 instr mpulse out endin 1 32000, 0.1, 0.05 a1
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK September, 2000 (New in Csound version 4.08)
Page 36-5
Page 36-6
37
SIGNAL GENERATORS: ADDITIVE S Y N T H E S I S /R E S Y N T H E S I S

adsyn
adsyn kamod, kfmod, ksmod, ifilcod
37.1
ar
DESCRIPTION
Output is an additive set of individually controlled sinusoids using an oscillator bank.
ifilcod integer or character-string denoting a control-file derived from analysis of an audio signal. An integer denotes the suffix of a file adsyn.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not fullpath, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined). adsyn control contains breakpoint amplitude and frequency envelope values organized for oscillator resynthesis. Memory usage depends on the size of the file involved, which are read and held entirely in memory during computation but are shared by multiple calls.
PERFORMANCE
adsyn synthesizes complex time-varying timbres through the method of additive synthesis. Any number of sinusoids, each individually controlled in frequency and amplitude, can be summed by high-speed arithmetic to produce a high-fidelity result. Component sinusoids are described by a control file describing amplitude and frequency tracks in millisecond breakpoint fashion. Tracks are defined by sequences of 16-bit binary integers:
-1, time, amp, time, amp,... -2, time, freq, time, freq,...
such as from hetrodyne filter analysis of an audio file. (for details see hetro.) The instantaneous amplitude and frequency values are used by an internal fixed-point oscillator that adds each active partial into an accumulated output signal. While there is a practical limit (limit removed in version 3.47) on the number of contributing partials, there is no restriction on their behavior over time. Any sound that can be described in terms of the behavior of sinusoids can be synthesized by adsyn alone. Sound described by an adsyn control file can also be modified during re-synthesis. The signals kamod, kfmod, ksmod will modify the amplitude, frequency, and speed of contributing partials. These are multiplying factors, with kfmod modifying the frequency and ksmod modifying the speed with which the millisecond breakpoint line-segments are traversed. Thus .7, 1.5, and 2 will give rise to a softer sound, a perfect fifth higher, but only half as long. The values 1,1,1 will leave the sound unmodified. Each of these inputs can be a control signal.
Signal Generators: Additive Synthesis/Resynthesis
Page 37-1
kfmod is a control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and .5 down an octave.
Page 37-2
37.2
ar
adsynt
adsynt kamp, kcps, iwfn, ifreqfn, iampfn, icnt[, iphs]
DESCRIPTION
Performs additive synthesis with an arbitrary number of partials, not necessarily harmonic.
iwfn table containing a waveform, usually a sine. Table values are not interpolated for performance reasons, so larger tables provide better quality. ifreqfn table containing frequency values for each partial. ifreqfn may contain beginning frequency values for each partial, but is usually used for generating parameters at runtime with tablew. Frequencies must be relative to kcps. Size must be at least icnt. iampfn table containing amplitude values for each partial. iampfn may contain beginning amplitude values for each partial, but is usually used for generating parameters at runtime with tablew. Amplitudes must be relative to kamp. Size must be at least icnt. icnt number of partials to be generated iphs initial phase of each oscillator, if iphs = -1, initialization is skipped. If iphs > 1, all phases will be initialized with a random value.
PERFORMANCE
kamp amplitude of note kcps base frequency of note. Partial frequencies will be relative to kcps. Frequency and amplitude of each partial is given in the two tables provided. The purpose of this opcode is to have an instrument generate synthesis parameters at k-rate and write them to global parameter tables with the tablew opcode.
E X A M PL E S
These two instruments perform additive synthesis. The output of each sounds like a Tibetan bowl. The first one is static, as parameters are only generated at init-time. In the second one, parameters are continuously changed.
gifrqs giamps ftgen ftgen instr 1 = = 2, 0, 32, 7, 0, 32, 0 ; for adsynt 3, 0, 32, 7, 0, 32, 0 ; parameters 10 0 ; generate two emty tables ; for freqency and amp
icnt index
; generates parameters at init time ; generate 10 voices ; init loop index
Page 37-3
loop: ifreq iamp index if asig endin
; loop only executed at ; init time index + 1, 1.5 ; define non-harmonic partials pow = 1 / (index+1) ; define amplitudes tableiw ifreq, index, gifrqs ; write to tables tableiw iamp, index, giamps ; used by adsynt tableiw = index + 1 (index < icnt) igoto loop ; do loop adsynt out 5000, 150, giwave, gifrqs, giamps, icnt asig
instr 2 icnt kindex loop: kspeed kphas voice klfo kdepth kfreq kfreq = = pow phasorbnk table pow pow = tablew pow phasorbnk table pow = tablew
; generates paramaters ; every k-cycle 10 ; generate 10 voices 0 ; reset loop index ; loop executed every ; k-cycle kindex + 1, 1.6 ; generate lfo for ; frequencies kspeed * 0.7, kindex, icnt ; individual phase for each kphas, giwave, 1 1.4, kindex ; arbitrary parameter twiddling... kindex + 1, 1.5 kfreq + klfo*0.006*kdepth kfreq, kindex, gifrqs ; write freqs to table for ; adsynt kindex + 1, 0.8 ; generate lfo for amplitudes kspeed*0.13, kindex, icnt, 2 ; individual phase for ; each voice kphas, giwave, 1 1 / (kindex + 1), 0.4 ; arbitrary parameter ; twiddling... kamp * (0.3+0.35*(klfo+1)) kamp, kindex, giamps ; write amps to table for ; adsynt
kspeed kphas klfo kamp kamp
kindex if giwave asig endin
= kindex + 1 (kindex < icnt) kgoto loop ; do loop 1, 0, 1024, 10, 1 ; generate a sinewave ftgen ; table 5000, 150, giwave, gifrqs, giamps, icnt adsynt asig out
AUTHOR
Page 37-4
37.3
ar
hsboscil
hsboscil kamp, ktone, kbrite, ibasfreq, iwfn, ioctfn \\ [, ioctcnt[, iphs]]
DESCRIPTION
An oscillator which takes tonality and brightness as arguments, relative to a base frequency.
ibasfreq base frequency to which tonality and brighness are relative iwfn function table of the waveform, usually a sine ioctfn function table used for weighting the octaves, usually something like:
f1 0 1024 -19 1 0.5 270 0.5
ioctcnt number of octaves used for brightness blending. Must be in the range 2 to 10. Default is 3. iphs initial phase of the oscillator. If iphs = -1, initialization is skipped.
PERFORMANCE
kamp amplitude of note ktone cyclic tonality parameter relative to ibasfreq in logarithmic octave, range 0 to 1, values > 1 can be used, and are internally reduced to frac(ktone). kbrite brightness parameter relative to ibasfreq, achieved by weighting ioctcnt octaves. It is scaled in such a way, that a value of 0 corresponds to the orignal value of ibasfreq, 1 corresponds to one octave above ibasfreq, -2 corresponds to two octaves below ibasfreq, etc. kbrite may be fractional. hsboscil takes tonality and brightness as arguments, relative to a base frequency (ibasfreq). Tonality is a cyclic parameter in the logarithmic octave, brightness is realized by mixing multiple weighted octaves. It is useful when tone space is understood in a concept of polar coordinates. Making ktone a line, and kbrite a constant, produces Rissets glissando. Oscillator table iwfn is always read interpolated. Performance time requires about ioctcnt * oscili.
Page 37-5
E X A M PL E S
giwave giblend ktona asig endin itona ibrite ibase kenv asig endin ftgen ftgen instr 1 line hsboscil out 1, 0, 1024, 10, 1, 1, 1, 1 2, 0, 1024, -19, 1, 0.5, 270, 0.5 ; synth wave ; blending window
; endless glissando 0,10,1 10000, ktona, 0, 200, giwave, giblend, 5 asig
; MIDI instrument: all octaves sound alike, instr 2 ; velocity is mapped to brightness octmidi 3 ampmidi = cpsoct(6) 20000, 1, 100 expon kenv, itona, ibrite, ibase, giwave, giblend, 5 hsboscil asig out
AUTHOR
Page 37-6
38
38.1
ar ar
S I G N A L G E N E R A T O R S : FM S Y N T H E S I S
foscil, foscili
foscil foscili xamp, kcps, xcar, xmod, kndx, ifn[, iphs] xamp, kcps, xcar, xmod, kndx, ifn[, iphs]
DESCRIPTION
Basic frequency modulated oscillators.
ifn function table number. Requires a wrap-around guard point. iphs (optional) initial phase of waveform in table ifn, expressed as a fraction of a cycle (0 to 1). A negative value will cause phase initialization to be skipped. The default value is 0.
38.1.1
PERFORMANCE foscil is a composite unit that effectively banks two oscils in the familiar Chowning FM setup, wherein the audio-rate output of one generator is used to modulate the frequency input of another (the carrier). Effective carrier frequency = kcps * kcar, and modulating frequency = kcps * xmod. For integral values of xcar and xmod, the perceived fundamental will be the minimum positive value of kcps * (xcar n * xmod), n = 1,1,2,... The input kndx is the index of modulation (usually time-varying and ranging 0 to 4 or so) which determines the spread of acoustic energy over the partial positions given by n = 0,1,2,.., etc. ifn should point to a stored sine wave. Previous to version 3.50, xcar and xmod could be k-rate only.
foscili differs from foscil in that the standard procedure of using a truncated phase as a sampling index is here replaced by a process that interpolates between two successive lookups. Interpolating generators will produce a noticeably cleaner output signal, but they may take as much as twice as long to run. Adequate accuracy can also be gained without the time cost of interpolation by using large stored function tables of 2K, 4K or 8K points if the space is available.
Signal Generators: FM Synthesis
Page 38-1
38.2
ar
fmvoice
fmvoice kamp, kfreq, kvowel, ktilt, kvibamt, kvibrate, ifn1, \\ ifn2, ifn3, ifn4, ivibfn
DESCRIPTION
FM Singing Voice Synthesis
ifn1, ifn2, ifn3,ifn3 -- Tables, usually of sinewaves.
PERFORMANCE
kamp Amplitude of note. kfreq Frequency of note played. kvowel -- the vowel being sung, in the range 0-64 ktilt -- the spectral tilt of the sound in the range 0 to 99 kvibamt -- Depth of vibrato kvibrate -- Rate of vibrato
E X A M PL E
k1 a1 line fmvoice 0, p3, 64 31129.60, 110, k1, 0, 0.005, 6, 1,1,1,1,1
AUTHOR
John ffitch (after Perry Cook) University of Bath, Codemist Ltd. Bath, UK New in Csound version 3.47
Page 38-2
38.3
fmbell, fmrhode, fmwurlie, fmmetal, fmb3, fmpercfl

fmbell fmrhode fmwurlie fmmetal fmb3 fmpercfl kamp, ifn3, kamp, ifn3, kamp, ifn3, kamp, ifn3, kamp, ifn3, kamp, ifn3, kfreq, kc1, ifn4, ivfn kfreq, kc1, ifn4, ivfn kfreq, kc1, ifn4, ivfn kfreq, kc1, ifn4, ivfn kfreq, kc1, ifn4, ivfn kfreq, kc1, ifn4, ivfn kc2, kvdepth, kvrate, ifn1, ifn2,\\ kc2, kvdepth, kvrate, ifn1, ifn2,\\ kc2, kvdepth, kvrate, ifn1, ifn2,\\ kc2, kvdepth, kvrate, ifn1, ifn2,\\ kc2, kvdepth, kvrate, ifn1, ifn2,\\ kc2, kvdepth, kvrate, ifn1, ifn2,\\
a1 a1 a1 a1 a1 a1
DESCRIPTION
A family of FM sounds, all using 4 basic oscillators and various architectures, as used in the TX81Z synthesizer.
All these opcodes take 5 tables for initialization. The first 4 are the basic inputs and the last is the low frequency oscillator (LFO) used for vibrato. The last table should usually be a sine wave. For the other opcodes the initial waves should be as in the table:
ifn1 ifn2 ifn3 ifn4 sinewave fwavblnk fwavblnk sinewave sinewave sinewave
sinewave sinewave sinewave fmbell sinewave sinewave sinewave fmrhode sinewave sinewave sinewave fmwurlie sinewave twopeaks twopeaks fmmetal sinewave sinewave sinewave fmb3 sinewave sinewave sinewave fmpercfl The sounds produced are then: Tubular Bell fmbell Fender Rhodes Electric Piano fmrhode Wurlitzer Electric Piano fmwurlie Heavy Metal fmmetal Hammond B3 organ fmb3 Percussive Flute fmpercfl
PERFORMANCE
kamp Amplitude of note. kfreq Frequency of note played.
Page 38-3
kc1, kc2 -- Controls for the synthesizer, as in the table:

kc1 fmbell fmrhode fmwurlie fmmetal fmb3 fmpercfl kc2 Mod index Mod index Mod index Total mod Total mod Total mod 1 1 1 index index index Crossfade Crossfade Crossfade Crossfade Crossfade Crossfade of of of of of of two two two two two two outputs outputs outputs modulators modulators modulators Algorithm 5 5 5 3 4 4
kvdepth -- Vibrator depth kvrate -- Vibrator rate
AUTHOR
Page 38-4
39
SIGNAL GENERATORS: SAMPLE PLAYBACK

loscil, loscil3
xamp, kcps, ifn[, ibas[,imod1,ibeg1,iend1 \\ [, imod2,ibeg2,iend2]]] xamp, kcps, ifn[, ibas[,imod1,ibeg1,iend1 \\ [, imod2,ibeg2,iend2]]]
39.1
ar[,ar2] loscil ar[,ar2] loscil3
DESCRIPTION
Read sampled sound (mono or stereo) from a table, with optional sustain and release looping.
ifn function table number, typically denoting an AIFF sampled sound segment with prescribed looping points. The source file may be mono or stereo. ibas (optional) base frequency in Hz of the recorded sound. This optionally overrides the frequency given in the AIFF file, but is required if the file did not contain one. The default value is 261.626 Hz, i.e. middle C. (New in Csound 4.03). imod1, imod2 (optional) play modes for the sustain and release loops. A value of 1 denotes normal looping, 2 denotes forward & backward looping, 0 denotes no looping. The default value (-1) will defer to the mode and the looping points given in the source file. ibeg1, iend1, ibeg2, iend2 (optional, dependent on mod1, mod2) begin and end points of the sustain and release loops. These are measured in sample frames from the beginning of the file, so will look the same whether the sound segment is monaural or stereo.
PERFORMANCE
loscil samples the ftable audio at a-rate determined by kcps, then multiplies the result by xamp. The sampling increment for kcps is dependent on the tables base-note frequency ibas, and is automatically adjusted if the orchestra sr value differs from that at which the source was recorded. In this unit, ftable is always sampled with interpolation. If sampling reaches the sustain loop endpoint and looping is in effect, the point of sampling will be modified and loscil will continue reading from within that loop segment. Once the instrument has received a turnoff signal (from the score or from a MIDI noteoff event), the next sustain endpoint encountered will be ignored and sampling will continue towards the release loop end-point, or towards the last sample (henceforth to zeros). loscil is the basic unit for building a sampling synthesizer. Given a sufficient set of recorded piano tones, for example, this unit can resample them to simulate the missing tones. Locating the sound source nearest a desired pitch can be done via table lookup. Once a sampling instrument has begun, its turnoff point may be unpredictable and require an external release envelope; this is often done by gating the sampled audio with linenr, which will extend the duration of a turned-off instrument by a specific period while it implements a decay.
Signal Generators: Sample Playback
Page 39-1
loscil3 is experimental. It is identical to loscil, except that it uses cubic interpolation. New in Csound version 3.50.
E X A M PL E
inum icps iamp ifno ibas kamp asig notnum cpsmidi ampmidi table table linenr loscil 3000, inum, inum, iamp, kamp, 1 2 ;notnum to choose an audio sample 3 0, .05, .01 ;at noteoff, extend by 50 ms. icps, ifno, cpsoct(ibas/12. + 3)
Page 39-2
39.2
ar ar
lposcil, lposcil3
lposcil lposcil3 kamp, kfreqratio, kloop, kend, ifn [,iphs] kamp, kfreqratio, kloop, kend, ifn [,iphs]
DESCRIPTION
Read sampled sound (mono or stereo) from a table, with optional sustain and release looping, and high precision. lposcil3 uses cubic interpolation.
ifn function table number iphs (optional) initial phase of sampling, expressed as a fraction of a cycle (0 to 1). The default value is 0.
PERFORMANCE
kamp amplitude kfreqratio multiply factor of table frequency (for example: 1 = original frequency, 1.5 = a fifth up , .5 = an octave down) kloop loop point (in samples) kend end loop point (in samples) lposcil (looping precise oscillator) allows varying at k-rate, the starting and ending point of a sample contained in a table (GEN01). This can be useful when reading a sampled loop of a wavetable, where repeat speed can be varied during the performance.
AUTHORS
Gabriel Maldonado (lposcil) Italy 1998 (New in Csound version 3.52) John ffitch (lposcil3) University of Bath/Codemist Ltd. Bath, UK February, 1999 (New in Csound version 3.52)
Page 39-3
39.3
sfload, sfplist, sfilist, sfpassign, sfpreset, sfplay, sfplaym, sfinstr, sfinstrm

sfload sfpassign sfpreset sfplist sfilist "filename" istartndx, ifilhandle iprog, ibank, ifilhandle, iprendx ifilhandle ifilhandle ivel, inotnum, xamp, xfreq, iprendx [, iflag] ivel, inotnum, xamp, xfreq, iprendx [, iflag] ivel, inotnum, xamp, xfreq, instrnum, ifilhandle [, iflag] ivel, inotnum, xamp, xfreq, instrnum, ifilhandle [, iflag]
ir ir
a1, a2 `a1 a1, a2 a1
sfplay sfplaym sfinstr sfinstrm
DESCRIPTION
Csound support for the SoundFont2 (SF2) sample file format. These opcodes allow management the sample-structure of SF2 files. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format can be found in the Appendix. Note that sfload, sfpassign, and sfpreset are placed in the header section of a Csound orchestra.
ir output to be used by other SF2 opcodes. For sfload, ir is ifilhandle. For sfpreset, ir is iprendx. filename name of the SF2 file, with its complete path. It must be typed within doublequotes. Use / to separate directories. This applies to DOS and Windows as well, where using a backslash will generate an error. ifilhandle unique number generated by sfload opcode to be used as an identifier for a SF2 file. Several SF2 files can be loaded and activated at the same time. istartndx starting index preset by the user in bulk preset assignments (see below). iprendx preset index iprog program number of a bank of presets in a SF2 file ibank number of a specific bank of a SF2 file ivel velocity value inotnum MIDI note number value iflag flag regarding the behavior of xfreq and inotnum instrnum number of an instrument of a SF2 file.
Page 39-4
PERFORMANCE
xamp amplitude correction factor xfreq frequency value or frequency multiplier, depending by iflag. When iflag = 0, xfreq is a multiplier of a the default frequency, assigned by SF2 preset to the inotenum value. When iflag = 1, xfreq is the absolute frequency of the output sound, in Hz. Default is 0. sfload loads an entire SF2 file into memory. It returns a file handle to be used by other opcodes. Several instances of sfload can placed in the header section of an orchestra, allowing use of more than one SF2 file in a single orchestra. sfplist prints a list of all presets of a previously loaded SF2 file to the console. sfilist prints a list of all instruments of a previously loaded SF2 file to the console. sfpassign assigns all presets of a previously loaded SF2 file to a sequence of progressive index numbers, to be used later with the opcodes sfplay and sfplaym. istartndx specifies the starting index number. Any number of sfpassign instances can be placed in the header section of an orchestra, each one assigning presets belonging to different SF2 files. The user must take care that preset index numbers of different SF2 files do not overlap. sfpreset assigns an existing preset of a previously loaded SF2 file to an index number, to be used later with the opcodes sfplay and sfplaym. The user must previously know the program and the bank numbers of the preset in order to fill the corresponding arguments. Any number of sfpreset instances can be placed in the header section of an orchestra, each one assigning a different preset belonging to the same (or different) SF2 file to different index numbers. sfplay plays a preset, generating a stereo sound. ivel does not directly affect the amplitude of the output, but informs sfplay about which sample should be chosen in multisample, velocity-split presets. When iflag = 0, inotnum sets the frequency of the output according to the MIDI note number used, and xfreq is used as a multiplier. When iflag = 1, the frequency of the output, is set directly by xfreq. This allows the user to use any kind of micro-tuning based scales. However, this method is designed to work correctly only with presets tuned to the default equal temperament. Attempts to use this method with a preset already having nonstandard tunings, or with drum-kit-based presets, could give unexpected results. Adjustment of the amplitude can be done by varying the xamp argument, which acts as a multiplier. Notice that both xamp and xfreq can use k-rate as well as a-rate signals. Both arguments must use variables of the same rate, or sfplay will not work correctly. iprendx must contain the number of a previously assigned preset, or Csound will crash. sfplaym is a mono version of sfplay. It should be used with mono preset, or with the stereo presets in which stereo output is not required. It is faster than sfplay. sfinstr plays an SF2 instrument instead of a preset (an SF2 instrument is the base of a preset layer). instrnum specifies the instrument number, and the user must be sure that the specified number belongs to an existing instrument of a determinate soundfont bank. Notice that both xamp and xfreq can operate at k-rate as well as a-rate, but both arguments must work at the same rate. sfinstrm plays is a mono version of sfinstr. This is the fastest opcode of the SF2 family. These opcodes only support the sample structure of SF2 files. The modulator structure of the SoundFont2 format is not supported in Csound. Any modulation or processing to the sample data is left to the Csound user, bypassing all restrictions forced by the SF2 standard.
Page 39-5
AUTHOR
Gabriel Maldonado Italy May, 2000 (New in Csound Version 4.06)
Page 39-6
40
SIGNAL GENERATORS: GRANULAR SYNTHESIS

fof, fof2
fof fof2 xamp, xfund, xform, iolaps, ifna, ifnb, xamp, xfund, xform, iolaps, ifna, ifnb, koct, kband, kris, kdur, kdec,\\ itotdur[, iphs[, ifmode]] koct, kband, kris, kdur, kdec,\\ itotdur, kphs, kgliss
40.1
ar ar
DESCRIPTION
Audio output is a succession of sinusoid bursts initiated at frequency xfund with a spectral peak at xform. For xfund above 25 Hz these bursts produce a speech-like formant with spectral characteristics determined by the k-input parameters. For lower fundamentals this generator provides a special form of granular synthesis. fof2 implements k-rate incremental indexing into ifna function with each successive burst.
iolaps number of preallocated spaces needed to hold overlapping burst data. Overlaps are frequency dependent, and the space required depends on the maximum value of xfund * kdur. Can be over-estimated at no computation cost. Uses less than 50 bytes of memory per iolap. ifna, ifnb- table numbers of two stored functions. The first is a sine table for sineburst synthesis (size of at least 4096 recommended). The second is a rise shape, used forwards and backwards to shape the sineburst rise and decay; this may be linear (GEN07) or perhaps a sigmoid (GEN19). itotdur total time during which this fof will be active. Normally set to p3. No new sineburst is created if it cannot complete its kdur within the remaining itotdur. iphs (optional) initial phase of the fundamental, expressed as a fraction of a cycle (0 to 1). The default value is 0. ifmode (optional) formant frequency mode. If zero, each sineburst keeps the xform frequency it was launched with. If non-zero, each is influenced by xform continuously. The default value is 0.
PERFORMANCE
xamp peak amplitude of each sineburst, observed at the true end of its rise pattern. The rise may exceed this value given a large bandwidth (say, Q < 10) and/or when the bursts are overlapping. xfund the fundamental frequency (in Hertz) of the impulses that create new sinebursts. xform the formant frequency, i.e. freq of the sinusoid burst induced by each xfund impulse. This frequency can be fixed for each burst or can vary continuously (see ifmode).
Signal Generators: Granular Synthesis
Page 40-1
koct octaviation index, normally zero. If greater than zero, lowers the effective xfund frequency by attenuating odd-numbered sinebursts. Whole numbers are full octaves, fractions transitional. kband the formant bandwidth (at -6dB), expressed in Hz. The bandwidth determines the rate of exponential decay throughout the sineburst, before the enveloping described below is applied. kris, kdur, kdec rise, overall duration, and decay times (in seconds) of the sinusoid burst. These values apply an enveloped duration to each burst, in similar fashion to a Csound linen generator but with rise and decay shapes derived from the ifnb input. kris inversely determines the skirtwidth (at -40 dB) of the induced formant region. kdur affects the density of sineburst overlaps, and thus the speed of computation. Typical values for vocal imitation are .003,.02,.007. In the fof2 implementation, kphs allows k-rate indexing of function table ifna with each successive burst, making it suitable for time-warping applications. Values of for kphs are normalized from 0 to 1, 1 being the end of the function table ifna. kgliss sets the end pitch of each grain relative to the initial pitch, in octaves. Thus kgliss = 2 means that the grain ends two octaves above its initial pitch, while kgliss = -5/3 has the grain ending a perfect major sixth below. Note: There are no optional parameters in fof2 Csounds fof generator is loosely based on Michael Clarkes C-coding of IRCAMs CHANT program (Xavier Rodet et al.). Each fof produces a single formant, and the output of four or more of these can be summed to produce a rich vocal imitation. fof synthesis is a special form of granular synthesis, and this implementation aids transformation between vocal imitation and granular textures. Computation speed depends on kdur, xfund, and the density of any overlaps.
Page 40-2
40.2
ar
fog
fog xamp, xdens, xtrans, xspd, koct, kband, kris, kdur,\\ kdec, iolaps, ifna, ifnb, itotdur[, iphs[, itmode]]
DESCRIPTION
Audio output is a succession of grains derived from data in a stored function table ifna. The local envelope of these grains and their timing is based on the model of fof synthesis and permits detailed control of the granular synthesis.
iolaps number of pre-located spaces needed to hold overlapping grain data. Overlaps are density dependent, and the space required depends on the maximum value of xdens* kdur. Can be over-estimated at no computation cost. Uses less than 50 bytes of memory per iolaps. ifna, ifnb table numbers of two stored functions. The first is the data used for granulation, usually from a soundfile (GEN01). The second is a rise shape, used forwards and backwards to shape the grain rise and decay; this is normally a sigmoid (GEN19) but may be linear (GEN07). itotdur total time during which this fog will be active. Normally set to p3. No new grain is created if it cannot complete its kdur within the remaining itotdur. iphs (optional) initial phase of the fundamental, expressed as a fraction of a cycle (0 to 1). The default value is 0. itmode (optional) transposition mode. If zero, each grain keeps the xtrans value it was launched with. if non-zero, each is influenced by xtrans continuously. The default value is 0.
PERFORMANCE
xamp amplitude factor. Amplitude is also dependent on the number of overlapping grains, the interaction of the rise shape (ifnb) and the exponential decay (kband), and the scaling of the grain waveform (ifna). The actual amplitude may therefore exceed xamp. xdens density. The frequency of grains per second. xtrans transposition factor. The rate at which data from the stored function table ifna is read within each grain. This has the effect of transposing the original material. A value of 1 produces the original pitch. Higher values transpose upwards, lower values downwards. Negative values result in the function table being read backwards. xspd speed. The rate at which successive grains advance through the stored function table ifna. xspd is in the form of an index (0 to 1) to ifna. This determines the movement of a pointer used as the starting point for reading data within each grain. (xtrans determines the rate at which data is read starting from this pointer.) koct octaviation index. The operation of this parameter is identical to that in fof. kband, kris, kdur, kdec grain envelope shape. These parameters determine the exponential decay (kband), and the rise (kris), overall duration (kdur,) and decay (kdec ) times of the grain envelope. Their operation is identical to that of the local envelope parameters in fof.
Page 40-3
The Csound fog generator is by Michael Clarke, extending his earlier work based on IRCAMs fof algorithm.
E X A M PL E
;p4 ;p5 ;p = transposition factor = speed factor = function table for grain data
;scaling to reflect sample rate and table length i1 = sr/ftlen(p6) a1 i1*p5 ;index for speed phasor a2 fog 5000, 100, p4, a1, 0, 0, , .01, .02, .01, 2, p6, 1, p3, 0, 1
AUTHOR
Michael Clark Huddersfield May 1997
Page 40-4
40.3
ar
grain
grain xamp, xpitch, xdens, kampoff, kpitchoff, kgdur, igfn,\\ iwfn, imgdur [, igrnd]
DESCRIPTION
Generates granular synthesis textures.
igfn The ftable number of the grain waveform. This can be just a sine wave or a sampled sound. iwfn Ftable number of the amplitude envelope used for the grains (see also GEN20). imgdur Maximum grain duration in seconds. This the biggest value to be assigned to kgdur. igrn (optional) if non-zero, turns off grain offset randomness. This means that all grains will begin reading from the beginning of the igfn table. If zero (the default), grains will start reading from random igfn table positions.
PERFORMANCE
xamp Amplitude of each grain. xpitch Grain pitch. To use the original frequency of the input sound, use the formula: sndsr / ftlen(igfn) where sndsr is the original sample rate of the igfn sound. xdens Density of grains measured in grains per second. If this is constant then the output is synchronous granular synthesis, very similar to fof. If xdens has a random element (like added noise), then the result is more like asynchronous granular synthesis. kampoff Maximum amplitude deviation from kamp. This means that the maximum amplitude a grain can have is kamp + kampoff and the minimum is kamp. If kampoff is set to zero then there is no random amplitude for each grain. kpitchoff Maximum pitch deviation from kpitch in Hz. Similar to kampoff. kgdur Grain duration in seconds. The maximum value for this should be declared in imgdur. If kgdur at any point becomes greater than imgdur, it will be truncated to imgdur. The grain generator is based primarily on work and writings of Barry Truax and Curtis Roads.
E X A M PL E
A texture with gradually shorter grains and wider amp and pitch spread
;;;;;;;;;;;;;;; graintest.orc instr 1 insnd = 10 ibasfrq = 32000 / ftlen(insnd) ; Use original sample rate of insnd file kamp expseg 8000, p3/2, 8000, p3/2, 16000 kpitch line ibasfrq, p3, ibasfrq * .8 kdens line 600, p3, 200 kaoff line 0, p3, 5000 kpoff line 0, p3, ibasfrq * .5 kgdur line .4, p3, .1 imaxgdur = .5
Page 40-5
ar grain kamp, kpitch, kdens, kaoff, kpoff, kgdur, insnd, 5, imaxgdur, 0.0 out ar endin ;;;;;;;;;;;;;;; graintest.sco f5 0 512 20 2 ; Hanning window f10 0 65536 1 Sound.wav 0 0 0 i1 0 10 e
AUTHOR
Paris Smaragdis MIT May 1997
Page 40-6
40.4
asig
granule
granule xamp, ivoice, iratio, imode, ithd, ifn, ipshift,\\ igskip, igskip_os, ilength, kgap, igap_os, kgsize,\\ igsize_os, iatt, idec [,iseed[,ipitch1[,ipitch2\\ [,ipitch3[,ipitch4[,ifnenv]]]]]]
DESCRIPTION
The granule unit generator is more complex than grain, but does add new possibilities. granule is a Csound unit generator which employs a wavetable as input to produce granularly synthesized audio output. Wavetable data may be generated by any of the GEN subroutines such as GEN01 which reads an audio data file into a wavetable. This enable a sampled sound to be used as the source for the grains. Up to 128 voices are implemented internally. The maximum number of voices can be increased by redefining the variable MAXVOICE in the grain4.h file. granule has a build-in random number generator to handle all the random offset parameters. Thresholding is also implemented to scan the source function table at initialization stage. This facilitates features such as skipping silence passage between sentences. The characteristics of the synthesis are controlled by 22 parameters. xamp is the amplitude of the output and it can be either audio rate or control rate variable.
PERFORMANCE
xamp amplitude. ivoice number of voices. iratio ratio of the speed of the gskip pointer relative to output audio sample rate. e.g. 0.5 will be half speed. imode +1 grain pointer move forward (same direction of the gskip pointer), -1 backward (oppose direction to the gskip pointer) or 0 for random. ithd threshold, if the sampled signal in the wavetable is smaller then ithd, it will be skipped. ifn function table number of sound source. ipshift pitch shift control. If ipshift is 0, pitch will be set randomly up and down an octave. If ipshift is 1, 2, 3 or 4, up to four different pitches can be set amount the number of voices defined in ivoice. The optional parameters ipitch1, ipitch2, ipitch3 and ipitch4 are used to quantify the pitch shifts. igskip initial skip from the beginning of the function table in sec. igskip_os gskip pointer random offset in sec, 0 will be no offset. ilength length of the table to be used starting from igskip in sec. kgap gap between grains in sec. igap_os gap random offset in % of the gap size, 0 gives no offset. kgsize grain size in sec. igsize_os grain size random offset in % of grain size, 0 gives no offset. iatt attack of the grain envelope in % of grain size.
Page 40-7
idec decade of the grain envelope in % of grain size. [iseed] optional, seed for the random number generator, default is 0.5. [ipitch1], [ipitch2], [ipitch3], [ipitch4]- optional, pitch shift parameter, used when ipshift is set to 1, 2, 3 or 4. Time scaling technique is used in pitch shift with linear interpolation between data points. Default value is 1, the original pitch.
E X A M PL E
; Orchestra file: sr = 44100 kr = 4410 ksmps = 10 nchnls = 2 instr 1 ; k1 linseg a1 granule a2 granule outs endin
0,0.5,1,(p3-p2-1),1,0.5,0 p4*k1,p5,p6,p7,p8,p9,p10,p11,p12,p13,p14,p15,\ p16,p17,p18,p19,p20,p21,p22,p23,p24 p4*k1,p5,p6,p7,p8,p9,p10,p11,p12,p13,p14,p15,\ p16,p17,p18,p19, p20+0.17,p21,p22,p23,p24 a1,a2
; Score file: ; f statement read sound file sine.aiff in the SFDIR ; directory into f-table 1 f1 0 524288 1 sine.aiff 1 0 i1 0 10 2000 64 0.5 0 0 1 4 0 0.005 10 0.01 50 0.02 50 30 30 0.39 \ 1 1.42 0.29 2 e
The above example reads a sound file called sine.aiff into wavetable number 1 with 524,288 samples. It generates 10 seconds of stereo audio output using the wavetable. In the orchestra file, all parameters required to control the synthesis are passed from the score file. A linseg function generator is used to generate an envelope with 0.5 second of linear attack and decay. Stereo effect is generated by using different seeds for the two granule function calls. In the example, 0.17 is added to p20 before passing into the second granule call to ensure that all of the random offset events are different from the first one. In the score file, the parameters are interpreted as:
p5 p6 p7 p8 p9 p10 p11 p13 p14 p16 p18 p20 p21 (ivoice) the number of voices is set to 64 (iratio) is set to 0.5, it scan the wavetable at half of the speed of the audio output rate (imode) is set to 0, the grain pointer only move forward (ithd) is set to 0, skipping the thresholding process (ifn) is set to 1, function table number 1 is used (ipshift) is set to 4, four different pitches are going to be generated (igskip) is set to 0 and p12 (igskip_os) is set to 0.005, no skipping into the wavetable and a 5 mSec random offset is used (ilength) is set to 10, 10 seconds of the wavetable is to be used (kgap) is set to 0.01 and p15 (igap_os) is set to 50, 10 mSec gap with 50% random offset is to be used (kgsize) is set to 0.02 and p17 (igsize_os) is set to 50, 20 mSec grain with 50% random offset is used (iatt) and p19 (idec) are set to 30, 30% of linear attack and decade is applied to the grain (iseed) seed for the random number generator is set to 0.39 - p 24 are pitches set to 1 which is the original pitch, 1.42 which is a 5th up, 0.29 which is a 7th down and finally 2 which is an octave up.
Page 40-8
AUTHOR
Allan Lee Belfast 1996
Page 40-9
40.5
ar[,ac]
sndwarp, sndwarpst
sndwarp xamp, xtimewarp, xresample, ifn1, ibeg,\\ iwsize, irandw, ioverlap, ifn2, itimemode
ar1,ar2[,ac1,ac2]
sndwarpst xamp, xtimewarp, xresample, ifn1, ibeg,\\ iwsize, irandw, ioverlap, ifn2, itimemode
DESCRIPTION
sndwarp reads sound samples from a table and applies time-stretching and/or pitch modification. Time and frequency modification are independent from one another. For example, a sound can be stretched in time while raising the pitch! The window size and overlap arguments are important to the result and should be experimented with. In general they should be as small as possible. For example, start with iwsize=sr/10 and ioverlap=15. Try irandw=iwsize*.2. If you can get away with less overlaps, the program will be faster. But too few may cause an audible flutter in the amplitude. The algorithm reacts differently depending upon the input sound and there are no fixed rules for the best use in all circumstances. But with proper tuning, excellent results can be achieved.
ifn1 the number of the table holding the sound samples which will be subjected to the sndwarp processing. GEN01 is the appropriate function generator to use to store the sound samples from a pre-existing soundfile. ibeg the time in seconds to begin reading in the table (or soundfile). When itimemode is non- zero, the value of xtimewarp is offset by ibeg. iwsize the window size in samples used in the time scaling algorithm. irandw the bandwidth of a random number generator. The random numbers will be added to iwsize. ioverlap determines the density of overlapping windows. ifn2 a function used to shape the window. It is usually used to create a ramp of some kind from zero at the beginning and back down to zero at the end of each window. Try using a half a sine (i.e.: f1 0 16384 9 .5 1 0) which works quite well. Other shapes can also be used.
PERFORMANCE
ar single channel of output from the sndwarp unit generator while ar1 and ar2 are the stereo (left and right) outputs from sndwarpst. sndwarp assumes that the function table holding the sampled signal is a mono one while sndwarpst assumes that it is stereo. This simply means that sndwarp will index the table by single-sample frame increments and sndwarpst will index the table by a two-sample frame increment. The user must be aware then that if a mono signal is used with sndwarpst or a stereo one with sndwarp, time and pitch will be altered accordingly. ac in sndwarp and ac1, ac2 in sndwarpst, are single layer (no overlaps), unwindowed versions of the time and/or pitch altered signal. They are supplied in order to be able to balance the amplitude of the signal output, which typically contains many overlapping and windowed versions of the signal, with a clean version of the time-scaled and pitch-shifted signal. The sndwarp process can cause noticeable changes in amplitude, (up and down), due to a time differential between the overlaps when time-shifting is being done. When used with a balance unit, ac, ac1, ac2 can greatly enhance the quality of sound. They are
Page 40-10
optional, but note that in sndwarpst they must both be present in the syntax (use both or neither). An example of how to use this is given below. xamp the value by which to scale the amplitude (see note on the use of this when using ac, ac1, ac2). xtimewarp determines how the input signal will be stretched or shrunk in time. There are two ways to use this argument depending upon the value given for itimemode. When the value of itimemode is 0, xitimewarp will scale the time of the sound. For example, a value of 2 will stretch the sound by 2 times. When itimemode is any non-zero value then xtimewarp is used as a time pointer in a similar way in which the time pointer works in lpread and pvoc. An example below illustrates this. In both cases, the pitch will not be altered by this process. Pitch shifting is done independently using xresample. xresample the factor by which to change the pitch of the sound. For example, a value of 2 will produce a sound one octave higher than the original. The timing of the sound, however, will not be altered.
E X A M PL E
The below example shows a slowing down or stretching of the sound stored in the stored table (ifn1). Over the duration of the note, the stretching will grow from no change from the original to a sound which is ten times slower than the original. At the same time the overall pitch will move upward over the duration by an octave.
iwindfun=1 isampfun=2 ibeg=0 iwindsize=2000 iwindrand=400 ioverlap=10 awarp line aresamp line kenv line asig sndwarp
1, p3, 10 1, p3, 2 1, p3, .1 kenv,awarp,aresamp,isampfun,ibeg,iwindsize,iwindrand, \\ ioverlap,iwindfun,0
Now, heres an example using xtimewarp as a time pointer and using stereo:
itimemode atime ar1, ar2 1 = 0, p3, 10 line sndwarpst kenv, atime, aresamp, sampfun, ibeg, \\ iwindsize, iwindrand, ioverlap, \\ iwindfun, itimemode
In the above, atime advances the time pointer used in the sndwarp from 0 to 10 over the duration of the note. If p3 is 20 then the sound will be two times slower than the original. Of course you can use a more complex function than just a single straight line to control the time factor. Now the same as above but using the balance function with the optional outputs:
asig,acmp abal sndwarp 1,awarp,aresamp,isampfun,ibeg,iwindsize,iwindrand,\\ ioverlap,iwindfun,itimemode balance asig, acmp
asig1,asig2,acmp1,acmp2 sndwarpst 1, atime, aresamp, sampfun,\\ ibeg, iwindsize, iwindrand, \\ ioverlap, iwindfun, itimemode abal1 balance asig1, acmp1 abal2 balance asig2, acmp2
In the above two examples notice the use of the balance unit. The output of balance can then be scaled, enveloped, sent to an out or outs, and so on. Notice that the amplitude arguments to sndwarp and sndwarpst are 1 in these examples. By scaling the signal after the sndwarp process, abal, abal1, and abal2 should contain signals that have nearly
Page 40-11
the same amplitude as the original input signal to the sndwarp process. This makes it much easier to predict the levels and avoid samples out of range or sample values that are too small. More advice: Only use the stereo version when you really need to be processing a stereo file. It is somewhat slower than the mono version and if you use the balance function it is slower again. There is nothing wrong with using a mono sndwarp in a stereo orchestra and sending the result to one or both channels of the stereo output!
AUTHOR
Richard Karpen Seattle, Wash 1997
Page 40-12
41
SIGNAL GENERATORS: SCANNED SYNTHESIS

Scanned synthesis is a variant of physical modeling, where a network of masses connected by springs is used to generate a dynamic waveform. The opcode scanu defines the mass/spring network and sets it in motion. The opcode scans follows a predefined path (trajectory) around the network and outputs the detected waveform. Several scans instances may follow different paths around the same network. These are highly efficient mechanical modelling algorithms for both synthesis and sonic animation via algorithmic processing. They should run in real-time. Thus, the output is useful either directly as audio, or as controller values for other parameters. The Csound implementation adds support for a scanning path or matrix. Essentially, this offers the possibility of reconnecting the masses in different orders, causing the signal to propagate quite differently. They do not necessarily need to be connected to their direct neighbors. Essentially, the matrix has the effect of molding this surface into a radically different shape. To produce the matrices, the table format is straightforward. For example, for 4 masses we have the following grid describing the possible connections:
Whenever two masses are connected, the point they define is 1. If two masses are not connected, then the point they define is 0. For example, a unidirectional string has the following connections: (1,2), (2,3), (3,4). If it is bidirectional, it also has (2,1), (3,2), (4,3)). For the unidirectional string, the matrix appears:
The above table format of the connection matrix is for conceptual convenience only. The actual values shown in te table are obtained by scans from an ASCII file using GEN23. The actual ASCII file is created from the table model row by row. Therefore the ASCII file for the example table shown above becomes:
0100001000010000
Signal Generators: Scanned Synthesis
Page 41-1
This marix example is very small and simple. In practice, most scanned synthesis instruments will use many more masses than four, so their matrices will be much larger and more complex. See the example in the scans documentation. Please note that the generated dynamic wavetables are very unstable. Certain values for masses, centering, and damping can cause the system to blow up and the most interesting sounds to emerge from your loudspeakers! The supplement to this manual contains a tutorial on scanned synthesis. The tutorial, examples, and other information on scanned synthesis is available from the Scanned Synthesis page at cSounds.com (http://www.csounds.com/scanned). Scanned synthesis developed by Bill Verplank, Max Mathews and Rob Shaw at Interval Research between 1998 and 2000.
Page 41-2
41.1
scanu
scanu init, irate, ifnvel, ifnmass, ifnstif, ifncentr, ifndamp, kmass, kstif, kcentr, kdamp, ileft, iright, kpos, kstrngth, ain, idisp, id
DESCRIPTION
Compute the waveform and the wavetable for use in scanned synthesis.
init the initial position of the masses. If this is a negative number, then the absolute of init signifies the table to use as a hammer shape. If init > 0, the length of it should be the same as the intended mass number, otherwise it can be anything. ifnvel the ftable that contains the initial velocity for each mass. It should have the same size as the intended mass number. ifnmass ftable that contains the mass of each mass. It should have the same size as the intended mass number. ifnstif ftable that contains the spring stiffness of each connection. It should have the same size as the square of the intended mass number. The data ordering is a row after row dump of the connection matrix of the system. ifncentr ftable that contains the centering force of each mass. It should have the same size as the intended mass number. ifndamp the ftable that contains the damping factor of each mass. It should have the same size as the intended mass number. ileft If init < 0, the position of the left hammer (ileft = 0 is hit at leftmost, ileft = 1 is hit at rightmost). iright If init < 0, the position of the right hammer (iright = 0 is hit at leftmost, iright = 1 is hit at rightmost). idisp If 0, no display of the masses is provided. id If positive, the ID of the opcode. This will be used to point the scanning opcode to the proper waveform maker. If this value is negative, the absolute of this value is the wavetable on which to write the waveshape. That wavetable can be used later from an other opcode to generate sound. The initial contents of this table will be destroyed.
PERFORMANCE
kmass scales the masses kstif scales the spring stiffness kcentr scales the centering force kdamp scales the damping kpos position of an active hammer along the string (kpos = 0 is leftmost, kpos = 1 is rightmost). The shape of the hammer is determined by init and the power it pushes with is kstrngth. kstrngth power that the active hammer uses
Page 41-3
ain audio input that adds to the velocity of the masses. Amplitude should not be too great.
E X A M PL E
For an example, see the documentation on scans.
AUTHOR
Paris Smaragdis MIT Media Lab Boston, Massachussetts USA March, 2000 (New in Csound version 4.05)
Page 41-4
41.2
ar
scans
scans kamp, kfreq, ifn, id[, iorder]
DESCRIPTION
Generate audio output using scanned synthesis.
ifn ftable containing the scanning trajectory. This is a series of numbers that contains addresses of masses. The order of these addresses is used as the scan path. It should not contain values greater than the number of masses, or negative numbers. See the introduction to the scanned synthesis section. id ID number of the scanu opcode's waveform to use iorder (optional) order of interpolation used internally. It can take any value in the range 1 to 4, and defaults to 4, which is quartic interpolation. The setting of 2 is quadratic and 1 is linear. The higher numbers are slower, but not necessarily better.
PERFORMANCE
kamp output amplitude. Note that the resulting amplitude is also dependent on instantaneous value in the wavetable. This number is effectively the scaling factor of the wavetable. kfreq frequency of the scan rate
E X A M PL E
Here is a simple example of scanned synthesis. The user must supply the matrix file "string128". This file, as well as several other matrices, is available in a zipped file from the Scanned Synthesis page at cSounds.com (http://www.csounds.com/scanned).
;orchestra -----------------------------------------= 44100 sr = 4410 kr = 10 ksmps = 1 nchnls a0 1, 2 a1 instr = scanu scans out endin 1 0 1,.01, 6, 2, 3, 4, 5, 2, .1, .1, -.01, .1, .5, 0, 0, a0, ampdb(p4), cpspch(p5), 7, a1 2
;score ---------------------------------------------; Initial condition f1 0 128 7 0 64 1 64 0 ; Masses f2 0 128 -7 1 128 1 ; Spring matrices f3 0 16384 -23 "string-128" ; Centering force f4 0 128 -7 0 128 2
Page 41-5
; Damping f5 0 128 -7 1 128 1 ; Initial velocity f6 0 128 -7 0 128 0 ; Trajectories f7 0 128 -5 .001 128 128 ; Note list i1 0 10 86 6.00 i1 11 14 86 7.00 i1 15 20 86 5.00 e
AUTHOR
Paris Smaragdis MIT Media Lab Boston, Massachussetts USA March, 2000 (New in Csound version 4.05)
Page 41-6
42
SIGNAL GENERATORS: WAVEGUIDE PHYSICAL MODELING

pluck
pluck kamp, kcps, icps, ifn, imeth [, iparm1, iparm2]
42.1
ar
DESCRIPTION
Audio output is a naturally decaying plucked string or drum sound based on the KarplusStrong algorithms.
icps intended pitch value in Hz, used to set up a buffer of 1 cycle of audio samples which will be smoothed over time by a chosen decay method. icps normally anticipates the value of kcps, but may be set artificially high or low to influence the size of the sample buffer. ifn table number of a stored function used to initialize the cyclic decay buffer. If ifn = 0, a random sequence will be used instead. imeth method of natural decay. There are six, some of which use parameters values that follow. 1. simple averaging. A simple smoothing process, uninfluenced by parameter values. 2. stretched averaging. As above, with smoothing time stretched by a factor of iparm1 ( = 1 ). 3. simple drum. The range from pitch to noise is controlled by a roughness factor in iparm1 (0 to 1). Zero gives the plucked string effect, while 1 reverses the polarity of every sample (octave down, odd harmonics). The setting .5 gives an optimum snare drum. 4. stretched drum. Combines both roughness and stretch factors. iparm1 is roughness (0 to 1), and iparm2 the stretch factor ( = 1 ). 5. weighted averaging. As method 1, with iparm1 weighting the current sample (the status quo) and iparm2 weighting the previous adjacent one. iparm1 + iparm2must be <= 1. 6. 1st order recursive filter, with coefs .5. Unaffected by parameter values. iparm1, iparm2 (optional) parameter values for use by the smoothing algorithms (above). The default values are both 0.
PERFORMANCE
An internal audio buffer, filled at i-time according to ifn, is continually resampled with periodicity kcps and the resulting output is multiplied by kamp. Parallel with the sampling, the buffer is smoothed to simulate the effect of natural decay. Plucked strings (1,2,5,6) are best realized by starting with a random noise source, which is rich in initial harmonics. Drum sounds (methods 3,4) work best with a flat source (wide pulse), which produces a deep noise attack and sharp decay. The original Karplus-Strong algorithm used a fixed number of samples per cycle, which caused serious quantization of the pitches available and their intonation. This
Signal Generators: Waveguide Physical Modeling
Page 42-1
implementation resamples a buffer at the exact pitch given by kcps, which can be varied for vibrato and glissando effects. For low values of the orch sampling rate (e.g. sr = 10000), high frequencies will store only very few samples (sr / icps). Since this may cause noticeable noise in the resampling process, the internal buffer has a minimum size of 64 samples. This can be further enlarged by setting icps to some artificially lower pitch.
Page 42-2
42.2
ar
wgpluck
wgpluck icps, iamp, kpick, iplk, idamp, ifilt, axcite
DESCRIPTION
A high fidelity simulation of a plucked string, using interpolating delay-lines.
icps frequency of plucked string iamp amplitude of string pluck iplk point along the string, where it is plucked, in the range of 0 to 1. 0 = no pluck idamp damping of the note. This controls the overall decay of the string. The greater the value of idamp1, the faster the decay. Negative values will cause an increase in output over time. ifilt control the attenuation of the filter at the bridge. Higher values cause the higher harmonics to decay faster.
PERFORMANCE
kpick proportion of the way along the point to sample the output axcite signal which excites the string A string of frequency icps is plucked with amplitude iamp at point iplk. The decay of the virtual string is controlled by idamp and ifilt which simulate the bridge. The oscillation is sampled at the point kpick, and excited by the signal axcite.
E X A M PL E S
The following example produces a moderately long note with rapidly decaying upper partials:
instr 1 1, 1, 1 oscil 220, 120, .5, 0, 10, 1000, axcite wgpluck apluck out endin whereas the following produces a shorter, brighter note: instr 1 axcite 1, 1, 1 oscil apluck wgpluck 220, 120, .5, 0, 30, 10, axcite apluck out endin axcite apluck
Page 42-3
42.3
ar ar
repluck, wgpluck2
wgpluck2 repluck iplk, xam, icps, kpick, krefl iplk, xam, icps, kpick, krefl, axcite
DESCRIPTION
wgpluck2 is an implementation of the physical model of the plucked string, with control over the pluck point, the pickup point and the filter. repluck is the same operation, but with an additional audio signal, axcite, used to excite the string. Both opcodes are based on the Karplus-Strong algorithms.
icps The string plays at icps pitch. iplck The point of pluck is iplk, which is a fraction of the way up the string (0 to 1). A pluck point of zero means no initial pluck.
PERFORMANCE
xamp Amplitude of note. kpick Proportion of the way along the string to sample the output. kabsp absorption coefficient at the bridge where 1 means total absorption and 0 is no absorption.
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK 1997
Page 42-4
42.4
ar
wgbow
wgbow kamp, kfreq, kpres, krat, kvibf, kvamp, ifn[, iminfreq]
DESCRIPTION
Audio output is a tone similar to a bowed string, using a physical model developed from Perry Cook, but re-coded for Csound.
ifn table of shape of vibrato, usually a sine table, created by a function iminfreq lowest frequency at which the instrument will play. If it is omitted it is taken to be the same as the initial kfreq. If iminfreq is negative, initialization will be skipped.
PERFORMANCE
A note is played on a string-like instrument, with the arguments as below. kamp Amplitude of note. kfreq Frequency of note played. kpres a parameter controlling the pressure of the bow on the string. Values should be about 3. The useful range is approximately 1 to 5. krat the position of the bow along the string. Usual playing is about 0.127236. The suggested range is 0.025 to 0.23. kvibf frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp amplitude of the vibrato
E X A M PL E
kv a1 linseg wgbow out 0, 0.5, 0, 1, 1, p3-0.5, 1 31129.60, 440, 3.0, 0.127236, 6.12723, kv*0.01, 1 a1
AUTHOR
Page 42-5
42.5
ar
wgflute
wgflute kamp, kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp,\\ ifn[, iminfreq[, kjetrf[, kendrf]]]
DESCRIPTION
Audio output is a tone similar to a flute, using a physical model developed from Perry Cook, but re-coded for Csound.
iatt time in seconds to reach full blowing pressure. 0.1 seems to correspond to reasonable playing. idetk time in seconds taken to stop blowing. 0.1 is a smooth ending ifn table of shape of vibrato, usually a sine table, created by a function iminfreq lowest frequency at which the instrument will play. If it is omitted it is taken to be the same as the initial kfreq. If iminfreq is negative, initialization will be skipped.
PERFORMANCE
kamp Amplitude of note. kfreq Frequency of note played. While it can be varied in performance, I have not tried it. kjet a parameter controlling the air jet. Values should be positive, and about 0.3. The useful range is approximately 0.08 to 0.56. kngain amplitude of the noise component, about 0 to 0.5 kvibf frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp amplitude of the vibrato kjetrf amount of reflection in the breath jet that powers the flute. Default value is 0.5. kendrf reflection coefficient of the breath jet. Default value is 0.5. Both ijetrf and iendrf are used in the calculation of the pressure differential.
E X A M PL E
a1 wgflute out 31129.60, 440, 0.32, 0.1, 0.1, 0.15, 5.925, 0.05, 1 a1
AUTHOR
Page 42-6
42.6
ar
wgbrass
wgbrass kamp, kfreq, ktens, iatt, kvibf, kvamp, ifn[, iminfreq]
DESCRIPTION
Audio output is a tone related to a brass instrument, using a physical model developed from Perry Cook, but re-coded for Csound. [NOTE: This is rather poor, and at present uncontrolled. Needs revision, and possibly more parameters].
iatt -- time taken to reach full pressure ifn table of shape of vibrato, usually a sine table, created by a function iminfreq lowest frequency at which the instrument will play. If it is omitted it is taken to be the same as the initial kfreq. If iminfreq is negative, initialization will be skipped.
PERFORMANCE
A note is played on a brass-like instrument, with the arguments as below. kamp Amplitude of note. kfreq Frequency of note played. ktens lip tension of the player. Suggested value is about 0.4 kvibf frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp amplitude of the vibrato
E X A M PL E
a1 wgbrass out 31129.60, 440, 0.1, 6.137, 0.05, 1 a1
AUTHOR
Page 42-7
42.7
ar
wgclar
wgclar kamp, kfreq, kstiff, iatt, idetk, kngain, kvibf, \\ kvamp, ifn[, iminfreq]
DESCRIPTION
Audio output is a tone similar to a clarinet, using a physical model developed from Perry Cook, but re-coded for Csound.
iatt time in seconds to reach full blowing pressure. 0.1 seems to correspond to reasonable playing. A longer time gives a definite initial wind sound. idetk time in seconds taken to stop blowing. 0.1 is a smooth ending ifn table of shape of vibrato, usually a sine table, created by a function iminfreq lowest frequency at which the instrument will play. If it is omitted it is taken to be the same as the initial kfreq. If iminfreq is negative, initialization will be skipped.
PERFORMANCE
A note is played on a clarinet-like instrument, with the arguments as below. kamp Amplitude of note. kfreq Frequency of note played. kstiff a stiffness parameter for the reed. Values should be negative, and about -0.3. The useful range is approximately -0.44 to -0.18. kngain amplitude of the noise component, about 0 to 0.5 kvibf frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp amplitude of the vibrato
E X A M PL E
a1 wgclar out 31129.60, 440, -0.3, 0.1, 0.1, 0.2, 5.735, 0.1, 1 a1
AUTHOR
Page 42-8
42.8
ar
wgbowedbar
wgbowedbar kamp, kfreq, kpos, kbowpres, kgain[, kconst, ktvel, kbowpos, ilow]
DESCRIPTION
A physical model of a bowed bar, belonging to the Perry Cook family of waveguide instruments.
ilow (optional) lowest frequency required
PERFORMANCE
kamp amplitude of signal kfreq frequency of signal kpos position of the bow on the bar, in the range 0 to 1 kbowpres pressure of the bow (as in wgbowed) kgain gain of filter. A value of about 0.809 is suggested. kconst (optional) an integration constant. Default is zero. ktvel (optional) either 0 or 1. When ktvel = 0, the bow velocity follows an ADSR style trajectory. When ktvel = 1, the value of the bow velocity decays in an exponentially. kbowpos (optional) the position on the bow, which affects the bow velocity trajectory.
E X A M PL E
;orchestra ---------------; ; ; ; ; ; kb kp kc a1 instr 1 pos bowpress gain intr trackvel bowpos line line line wgbowedbar out endin ;score ------------------i1 e 0 3 32000 7.00 0 = = = = = = [0, 1] [1, 10] [0.8, 1] [0,1] [0, 1] [0, 1]
0.5, p3, 0.1 0.6, p3, 0.7 1, p3, 1 p4, cpspch cpspch(p5), kb, kp, 0.995, p6, 0, kc, 50 a1
Page 42-9
AUTHOR
Page 42-10
43
SIGNAL GENERATORS: MODELS EMULATIONS

moog
moog
AND
43.1
a1
kamp, kfreq, kfiltq, kfiltrate, kvibf, kvamp, iafn,\\ iwfn, ivfn
DESCRIPTION
An emulation of a mini-Moog synthesizer.
iafn, iwfn, ivfn three table numbers containing the attack waveform (unlooped), the main looping wave form, and the vibrato waveform. The files mandpluk.aiff and impuls20.aiff are suitable for the first two, and a sine wave for the last.
PERFORMANCE
kamp Amplitude of note. kfreq Frequency of note played. kfiltq Q of the filter, in the range 0.8 to 0.9 kfiltrate rate control for the filter in the range 0 to 0.0002 kvibf frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp amplitude of the vibrato
AUTHOR
Signal Generators: Models and Emulations
Page 43-1
43.2
ar
shaker
shaker kamp, kfreq, kbeans, kdamp, ktimes[, idecay]
DESCRIPTION
Audio output is a tone related to the shaking of a maraca or similar gourd instrument. The method is a physically inspired model developed from Perry Cook, but re-coded for Csound.
idecay If present indicates for how long at the end of the note the shaker is to be damped. The default value is zero.
PERFORMANCE
A note is played on a maraca-like instrument, with the arguments as below. kamp Amplitude of note. kfreq Frequency of note played. kbeans The number of beans in the gourd. A value of 8 seems suitable, kdamp -- The damping value of the shaker. Values of 0.98 to 1 seems suitable, with 0.99 a reasonable default. ktimes -- Number of times shaken. The argument knum was redundant, so was removed in version 3.49.
E X A M PL E
a1 shaker outs 31129.60, 440, 8, 0.999, 100, 0 a1, a1
AUTHOR
Page 43-2
43.3
ar ar
marimba, vibes
marimba vibes kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, \\ idec[, idoubles[, itriples]] kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, \\ idec
DESCRIPTION
Audio output is a tone related to the striking of a wooden or metal block as found in a marimba or vibraphone. The method is a physical model developed from Perry Cook, but re-coded for Csound.
ihrd the hardness of the stick used in the strike. A range of 0 to 1 is used. 0.5 is a suitable value. ipos where the block is hit, in the range 0 to 1. imp a table of the strike impulses. The file marmstk1.wav is a suitable function from measurements, and can be loaded with a GEN01 table. ivfn shape of vibrato, usually a sine table, created by a function idec time before end of note when damping is introduced idoubles percentage of double strikes. Default is 40%. itriples percentage of triple strikes. Default is 20%.
PERFORMANCE
kamp Amplitude of note. kfreq Frequency of note played. kvibf frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp amplitude of the vibrato
E X A M PL E
a1 a2 marimba vibes outs 31129.60, 440, 0.5, 0.561, 2, 6.0, 0.05, 1, 0.1 31129.60, 440, 0.5, 0.561, 2, 4.0, 0.2, 1, 0.1a1 a1, a2
Page 43-3
AUTHOR
Page 43-4
43.4
ar
mandol
mandol kamp, kfreq, kpluck, kdetune, kgain, ksize, ifn\\ [, iminfreq]
DESCRIPTION
An emulation of a mandolin.
ifn -- table number containing the pluck wave form. The file mandpluk.aiff is suitable for this. iminfreq -- Lowest frequency to be played on the note. If it is omitted it is taken to be the same as the initial kfreq.
PERFORMANCE
kamp Amplitude of note. kfreq Frequency of note played. kpluck The pluck position, in range 0 to 1. Suggest 0.4. kgain the loopgain of the model, in the range 0.97 to 1. kdetune The proportional detuning between the two strings. Suggested range 1 and 0.9. ksize The size of the body of the mandolin. Range 0 to 2.
AUTHOR
Page 43-5
43.5
ar
gogobel
gogobel kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn
DESCRIPTION
Audio output is a tone related to the striking of a cow bell or similar. The method is a physical model developed from Perry Cook, but re-coded for Csound.
ihrd -- the hardness of the stick used in the strike. A range of 0 to 1 is used. 0.5 is a suitable value. ipos -- where the block is hit, in the range 0 to 1. imp a table of the strike impulses. The file marmstk1.wav is a suitable function from measurements, and can be loaded with a GEN01 table. ivfn shape of vibrato, usually a sine table, created by a function.
PERFORMANCE
A note is played on a cowbell-like instrument, with the arguments as below. kamp Amplitude of note. kfreq Frequency of note played. kvibf frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp amplitude of the vibrato
E X A M PL E
a1 gogobel outs 31129.60, 440, p4, 0.561, 3, 6.0, 0.3, 1 a1, a2
N A M E C HA N G E
Prior to Csound version 3.52 (February, 1999), this opcode was called agogobel.
AUTHOR
Page 43-6
43.6
ar
voice
voice kamp, kfreq, kphoneme, kform, kvibf, kvamp, ifn, ivfn
D E S C R I PT I O N
An emulation of a human voice.
ifn, ivfn two table numbers containing the carrier wave form and the vibrato waveform. The files impuls20.aiff, ahh.aiff, eee.aiff, or ooo.aiff are suitable for the first of these, and a sine wave for the second. These files are available from: ftp://ftp.maths.bath.ac.uk/pub/dream/documentation/sounds/modelling /
PERFORMANCE
kamp Amplitude of note. kfreq Frequency of note played. It can be varied in performance. kphoneme an integer in the range 0 to 16, which select the formants for the sounds: eee,ihh,ehh,aaa, ahh,aww,ohh,uhh, uuu,ooo,rrr,lll, mmm,nnn,nng,ngg. At present the phonemes fff,sss,thh,shh, xxx,hee,hoo,hah, bbb,ddd,jjj,ggg, vvv,zzz,thz,zhh are not available (!) kform Gain on the phoneme. values 0.0 to 1.2 recommended. kvibf frequency of vibrato in Hertz. Suggested range is 0 to 12 kvamp amplitude of the vibrato
AUTHOR
Page 43-7
43.7
ax, ay, az
lorenz
lorenz ks, kr, kb, kh, ix, iy, iz, iskip
DESCRIPTION
Implements the Lorenz system of equations. The Lorenz system is a chaotic-dynamic system which was originally used to simulate the motion of a particle in convection currents and simplified weather systems. Small differences in initial conditions rapidly lead to diverging values. This is sometimes expressed as the butterfly effect. If a butterfly flaps its wings in Australia, it will have an effect on the weather in Alaska. This system is one of the milestones in the development of chaos theory. It is useful as a chaotic audio source or as a low frequency modulation source.
ix, iy, iz the initial coordinates of the particle iskip used to skip generated values. If iskip is set to 5, only every fifth value generated is output. This is useful in generating higher pitched tones.
PERFORMANCE
ksv the Prandtl number or sigma krv the Rayleigh number kbv the ratio of the length and width of the box in which the convection currents are generated kh the step size used in approximating the differential equation. This can be used to control the pitch of the systems. Values of .1-.001 are typical. The equations are approximated as follows: x = x + h*(s*(y - x)) y = y + h*(-x*z + r*x - y) z = z + h*(x*y - b*z) The historical values of these parameters are: ks = 10 kr = 28 kb = 8/3
E X A M PL E
ksv = p4 krv = p5 kbv = p6 ax, ay, az instr 20
lorenz endin R 28
ksv, krv, kbv, .01, .6, .6, .6, 1
;score ; start dur S i20 5 1 10 e
V 2.667
Page 43-8
AUTHOR
Hans Mikelson February 1999 (New in Csound version 3.53)
Page 43-9
43.8
planet
planet kmass1, kmass2, ksep, ix, iy, iz, ivx, \\ ivy, ivz, idelta, ifriction
ax, ay, az
DESCRIPTION
planet simulates a planet orbiting in a binary star system. The outputs are the x, y and z coordinates of the orbiting planet. It is possible for the planet to achieve escape velocity by a close encounter with a star. This makes this system somewhat unstable.
ix, iy, iz the initial x, y and z coordinates of the planet ivx, ivy, ivz the initial velocity vector components for the planet. idelta the step size used to approximate the differential equation. ifriction a value for friction, which can used to keep the system from blowing up
PERFORMANCE
kmass1 the mass of the first star kmass2 the mass of the second star ksep determines the distance between the two stars ax, ay, az the output x, y, and z coordinates of the planet
E X A M PL E
instr 1 idur iamp km1 km2 ksep ix iy iz ivx ivy ivz ih ifric kamp = = = = = = = = = = = = = linseg p3 p4 p5 p6 p7 p8 p9 p10 p11 p12 p13 p14 p15 0, .002, iamp, idur-.004, iamp, .002, 0 km1, km2, ksep, ix, iy, iz, ivx, ivy, ivz, ih, ifric ax*kamp, ay*kamp
ax,ay,az planet outs endin ; i1 i1 i1 i1 i1 i1 Sta 0 + . . . . Dur 1 . . . . . Amp 5000 . . . . . M1 .5 .5 .4 .3 .25 .2 M2 .35 0 .3 .3 .3 .5
Sep 2.2 0 2 2 2 2
X 0 0 0 0 0 0
Y .1 .1 .1 .1 .1 .1
Z 0 0 0 0 0 0
VX .5 .5 .5 .5 .5 .5
VY .6 .6 .6 .6 .6 .6
VZ -.1 -.1 -.1 -.1 -.1 -.1
h .5 .5 .5 .5 .5 .1
Frict -0.1 0.1 0.0 0.1 1.0 1.0
Page 43-10
AUTHOR
Hans Mikelson December 1998 New in Csound version 3.50
Page 43-11
43.9
ar ar ar ar ar
cabasa, crunch, sekere, sandpaper, stix

cabasa crunch sekere sandpaper stix iamp, iamp, iamp, iamp, iamp, idettack[, idettack[, idettack[, idettack[, idettack[, knum, knum, knum, knum, knum, kdamp, kdamp, kdamp, kdamp, kdamp, kmaxshake] kmaxshake] kmaxshake] kmaxshake] kmaxshake]
DESCRIPTION
Semi-physical models of various percussion sounds.
iamp Amplitude of output. Note: As these instruments are stochastic, this is only a approximation. idettack period of time over which all sound is stopped
PERFORMANCE
knum The number of beads, teeth, bells, timbrels, etc. If, zero the default value is: cabasa crunch sekere sandpaper stix = 512.0000 = 7.0000 = 64.0000 = 128.0000 = 30.0000
kdamp the damping factor of the instrument. The value is used as an adjustment around the defaults, with 1 being no damping. If zero, the default values are used. The defaults are: cabasa crunch sekere sandpaper stix = = = = = 0.9970 0.99806 0.9990 0.9990 0.9980
kmaxshake amount of energy to add back into the system. The value should be in range 0 to 1.
E X A M PL E
;orchestra --------------sr = 44100 = 4410 kr = 10 ksmps 1 nchnls = gknum init 0 ;initialize optional arguments gkdamp 0 ; for use with all instruments init gkmaxshake init 0 ;an example of a cabasa instr 01 a1 cabasa p4, 0.01, gknum, gkdamp, gkmaxshake a1 out endin ;an example of a crunch instr 02 a1 crunch p4, 0.01, gknum, gkdamp, gkmaxshake a1 out endin ;an example of a sekere instr 03 The Public Csound Reference Manual Version 4.10 Signal Generators: Models and Emulations Page 43-12
p4, 0.01, gknum, gkdamp, gkmaxshake sekere a1 out endin ;an example of sandpaper blocks instr 04 a1 2, p3, 2 ;preset amplitude increase line a2 p4, 0.01, gknum, gkdamp, gkmaxshake sandpaper a3 a1, a2 ;increase amplitude product a3 out endin ;an example of stix instr 05 a1 20, p3, 20 ;preset amplitude increase line a2 p4, 0.01, gknum, gkdamp, gkmaxshake stix a3 a1, a2 ;increase amplitude product a3 out endin ;score ------------------i1 0 1 26000 i2 2 1 26000 i3 4 1 26000 i4 6 1 26000 i5 8 1 26000 e
a1
AUTHOR
John ffitch University of Bath, Codemist Ltd. Bath, UK New in Csound version 4.07
Page 43-13
43.10 g u i r o , t a m b o u r i n e , b a m b o o ,
dripwater, sleighbells
ar ar ar ar ar iamp, idettack[, iamp, idettack[, kfreq2] iamp, idettack[, bamboo kfreq2] iamp, idettack[, dripwater kfreq2] sleighbells iamp, idettack[, kfreq2] guiro tambourine knum, kdamp, kmaxshake, kfreq, kfreq1] knum, kdamp, kmaxshake, kfreq, kfreq1, knum, kdamp, kmaxshake, kfreq, kfreq1, knum, kdamp, kmaxshake, kfreq, kfreq1, knum, kdamp, kmaxshake, kfreq, kfreq1,
DESCRIPTION
Semi-physical models of various percussion sounds.
iamp Amplitude of output. Note: As these instruments are stochastic, this is only a approximation. idettack period of time over which all sound is stopped
PERFORMANCE
knum The number of beads, teeth, bells, timbrels, etc. If, zero the default value is: guiro tambourine bamboo dripwater sleighbells = 128.0000 = 32.0000 = 1.2500 = 10.0000 = 32.0000
kdamp the damping factor of the instrument. The value is used as an adjustment around the defaults, with 1 being no damping. If zero, the default values are used. The defaults are: guiro tambourine bamboo dripwater sleighbells = = = = = 1.0000 0.9985 0.9999 0.9950 0.9994
kmaxshake amount of energy to add back into the system. The value should be in range 0 to 1. kfreq the main resonant frequency. The default values are: guiro tambourine bamboo dripwater sleighbells = = = = = 2500.0000 2300.0000 2800.0000 450.0000 2500.0000
Page 43-14
kfreq1 the first resonant frequency. The default values are: tambourine bamboo dripwater sleighbells = 5600.0000 = 2240.0000 = 600.0000 = 5300.0000
kfreq2 the second resonant frequency. The default values are: tambourine bamboo dripwater sleighbells = 8100.0000 = 3360.0000 = 750.0000 = 6500.0000
E X A M PL E
;orchestra --------------= 44100 sr = 4410 kr = 10 ksmps 1 nchnls = instr 01 a1 p4, 0.01 guiro a1 out endin instr 02 a1 tambourine p4, 0.01 a1 out endin instr 03 a1 p4, 0.01 bamboo a1 out endin instr 04 a1 5, p3, 5 line a2 p4, 0.01, 0, .9 dripwater a3 a1, a2 product a3 out endin instr 05 a1 sleighbells p4, 0.01 a1 out endin ;score ------------------i1 0 1 20000 i2 2 1 20000 i3 4 1 20000 i4 6 1 20000 i5 8 1 20000 e
;example of a guiro
;example of a tambourine
;example of bamboo
;example of a water drip ;preset an amplitude boost ;increase amplitude ;an example of sleighbells
AUTHOR
Page 43-15
Page 43-16
44
S I G N A L G E N E R A T O R S : STFT R E S Y N T H E S I S (V O C O D I N G )
pvoc, vpvoc
pvoc vpvoc ktimpnt, kfmod, ifilcod [, ispecwp, iextractmode,\\ ifreqlim, igatefn] ktimpnt, kfmod, ifile[, ispecwp[, ifn]]
44.1
ar ar
DESCRIPTION
Output is an additive set of individually controlled sinusoids, using phase vocoder resynthesis.
ifilcod integer or character-string denoting a control-file derived from analysis of an audio signal. An integer denotes the suffix of a file pvoc.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not fullpath, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined).pvoc control contains breakpoint amplitude and frequency envelope values organized for fft resynthesis. Memory usage depends on the size of the file involved, which is read and held entirely in memory during computation, but are shared by multiple calls (see also lpread). ispecwp (optional) if non-zero, attempts to preserve the spectral envelope while its frequency content is varied by kfmod. The default value is zero. iextractmode (optional) determines if spectral extraction will be carried out, and if so, whether components that have changes in frequency below ifreqlim or above ifreqlim will be discarded. A value for iextractmode of 1 will cause pvadd to synthesize only those components where the frequency difference between analysis frames is greater than ifreqlim. A value of 2 for iextractmode will cause pvadd to synthesize only those components where the frequency difference between frames is less than ifreqlim. The default values for iextractmode and ifreqlim are 0, in which case a simple resynthesis will be done. See examples under pvadd for how to use spectral extraction. igatefn is the number of a stored function which will be applied to the amplitudes of the analysis bins before resynthesis takes place. If igatefn is greater than 0, the amplitudes of each bin will be scaled by igatefn through a simple mapping process. First, the amplitudes of all of the bins in all of the frames in the entire analysis file are compared to determine the maximum amplitude value. This value is then used create normalized amplitudes as indeces into the stored function igatefn. The maximum amplitude will map to the last point in the function. An amplitude of 0 will map to the first point in the function. Values between 0 and 1 will map accordingly to points along the function table. See examples under pvadd for how to use amplitude gating.
Signal Generators: STFT Resynthesis (Vocoding)
Page 44-1
ifn (optional) optional function table containing control information for vpvoc. If ifn = 0, control is derived internally from a previous tableseg or tablexseg unit. Default is 0. (New in Csound version 3.59)
PERFORMANCE
pvoc implements signal reconstruction using an fft-based phase vocoder. The control data stems from a precomputed analysis file with a known frame rate. The passage of time through this file is specified by ktimpnt, which represents the time in seconds. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file. kfmod is a control-rate transposition factor: a value of 1 incurs no transposition, 1.5 transposes up a perfect fifth, and .5 down an octave. This implementation of pvoc was orignally written by Dan Ellis. It is based in part on the system of Mark Dolson, but the pre-analysis concept is new. The spectral extraction and amplitude gating (new in Csound version 3.56) were added by Richard Karpen based on functions in SoundHack by Tom Erbe. vpvoc is identical to pvoc except that it takes the result of a previous tableseg, tablexseg and uses the resulting function table (passed internally to the vpvoc), as an envelope over the magnitudes of the analysis data channels. Optionally, a table specified by ifn, may be used. The result is spectral enveloping. The function size used in the tableseg should be framesize/2, where framesize is the number of bins in the phase vocoder analysis file that is being used by the vpvoc. Each location in the table will be used to scale a single analysis bin. By using different functions for ifn1, ifn2, etc.. in the tableseg, the spectral envelope becomes a dynamically changing one. See also tableseg and tablexseg.
E X A M PL E
The following example using vpvoc, shows the use of functions such as
f 1 0 256 5 .001 128 1 128 .001 f 2 0 256 5 1 128 .001 128 1 f 3 0 256 7 1 256 1
to scale the amplitudes of the separate analysis bins.

ktime apv line tablexseg vpvoc 0, p3,3 ; time pointer, in seconds, into file 1, p3*.5, 2, p3*.5, 3 ktime,1, pvoc.file
The result would be a time-varying spectral envelope applied to the phase vocoder analysis data. Since this amplifies or attenuates the amount of signal at the frequencies that are paired with the amplitudes which are scaled by these functions, it has the effect of applying very accurate filters to the signal. In this example the first table would have the effect of a bandpass filter , gradually be band-rejected over half the notes duration, and then go towards no modification of the magnitudes over the second half.
AUTHORS
Dan Ellis (pvoc) Richard Karpen (vpvoc) Seattle, Washngton 1997
Page 44-2
44.2
pvread, pvbufread, pvinterp, pvcross, tableseg, tablexseg

ktimpnt, ifile, ibin ktimpnt, ifile ktimpnt, kfmod, ifile, kfreqscale1, kfreqscale2,\\ kampscale1, kampscale2, kfreqinterp, kampinterp ktimpnt, kfmod, ifile, kamp1, kamp2[, ispecwp] ifn1, idur1, ifn2[, idur2, ifn3[...]] ifn1, idur1, ifn2[, idur2, ifn3[...]]
kfr, kamp pvread pvbufread ar pvinterp ar pvcross tableseg tablexseg
DESCRIPTION
pvread reads from a pvoc file and returns the frequency and amplitude from a single analysis channel or bin. The returned values can be used anywhere else in the Csound instrument. For example, one can use them as arguments to an oscillator to synthesize a single component from an analyzed signal or a bank of pvreads can be used to resynthesize the analyzed sound using additive synthesis by passing the frequency and magnitude values to a bank of oscillators. pvbufread reads from a pvoc file and makes the retrieved data available to any following pvinterp and pvcross units that appear in an instrument before a subsequent pvbufread (just as lpread and lpreson work together). The data is passed internally and the unit has no output of its own. pvinterp and pvcross allow the interprocessing of two phase vocoder analysis files prior to the resynthesis which these units do also. Both of these units receive data from one of the files from a previously called pvbufread unit. The other file is read by the pvinterp and/or pvcross units. Since each of these units has its own time-pointer the analysis files can be read at different speeds and directions from one another. pvinterp does not allow for the use of the ispecwp process as with the pvoc and vpvoc units. pvinterp interpolates between the amplitudes and frequencies, on a bin by bin basis, of two phase vocoder analysis files (one from a previously called pvbufread unit and the other from within its own argument list), allowing for user defined transitions between analyzed sounds. It also allows for general scaling of the amplitudes and frequencies of each file separately before the interpolated values are calculated and sent to the resynthesis routines. The kfmod argument in pvinterp performs its frequency scaling on the frequency values after their derivation from the separate scaling and subsequent interpolation is performed so that this acts as an overall scaling value of the new frequency components. pvcross applies the amplitudes from one phase vocoder analysis file to the data from a second file and then performs the resynthesis. The data is passed, as described above, from a previously called pvbufread unit. The two k-rate amplitude arguments are used to scale the amplitudes of each files separately before they are added together and used in the resynthesis (see below for further explanation). The frequencies of the first file are not used at all in this process. This unit simply allows for cross-synthesis through the application of the amplitudes of the spectra of one signal to the frequencies of a second signal. Unlike pvinterp, pvcross does allow for the use of the ispecwp as in pvoc and vpvoc. tableseg and tablexseg are like linseg and expseg but interpolate between values in a stored function tables. The result is a new function table passed internally to any following vpvoc which occurs before a subsequent tableseg or tablexseg (much like lpread/lpreson pairs work). The uses of these are described below under vpvoc.
Page 44-3
ifile the pvoc number (n in pvoc.n) or the name in quotes of the analysis file made using pvanal. (See pvoc.) ibin the number of the analysis channel from which to return frequency in Hz and magnitude. ifn1, ifn2, ifn3, etc. function table numbers for tableseg and tablexseg. ifn1, ifn2, and so on, must be the same size. idur1, idur2, etc. durations for tableseg and tablexseg, during which interpolation from one table to the next will take place.
PERFORMANCE
kfreq, kamp outputs of the pvread unit. These values, retrieved from a phase vocoder analysis file, represent the values of frequency and amplitude from a single analysis channel specified in the ibin argument. Interpolation between analysis frames is performed at k-rate resolution and dependent of course upon the rate and direction of ktimpnt. ktimpnt, kfmod, ispecwp used for pvread exactly the same as for pvoc (see above description of pvinterp for its special use of kfmod). kfreqscale1, kfreqscale2, kampscale1, kampscale2 used in pvinterp to scale the frequencies and amplitudes stored in each frame of the phase vocoder analysis file. kfreqscale1 and kampscale1 scale the frequencies and amplitudes of the data from the file read by the previously called pvbufread (this data is passed internally to the pvinterp unit). kfreqscale2 and kampscale2 scale the frequencies and amplitudes of the file named by ifile in the pvinterp argument list and read within the pvinterp unit. By using these arguments it is possible to adjust these values before applying the interpolation. For example, if file1 is much louder than file2, it might be desirable to scale down the amplitudes of file1 or scale up those of file2 before interpolating. Likewise one can adjust the frequencies of each to bring them more in accord with one another (or just the opposite, of course!) before the interpolation is performed. kfreqinterp, kampinterp used in pvinterp to determine the interpolation distance between the values of one phase vocoder file and the values of a second file. When the value of kfreqinterp is 0, the frequency values will be entirely those from the first file (read by the pvbufread), post scaling by the kfreqscale1 argument. When the value of kfreqinterp is 1 the frequency values will be those of the second file (read by the pvinterp unit itself), post scaling by kfreqscale2. When kfreqinterp is between 0 and 1 the frequency values will be calculated, on a bin, by bin basis, as the percentage between each pair of frequencies (in other words, kfreqinterp=.5 will cause the frequencies values to be half way between the values in the set of data from the first file and the set of data from the second file). kampinterp1 and kampinterp2 work in the same way upon the amplitudes of the two files. Since these are k-rate arguments, the percentages can change over time making it possible to create many kinds of transitions between sounds.
E X A M PL E
The example below shows the use pvread to synthesize a single component from a phase vocoder analysis file. It should be noted that the kfreq and kamp outputs can be used for any kind of synthesis, filtering, processing, and so on.
ktime krefq,kamp asig line pvread oscili 0, p3, 3 ktime, pvoc.file, 7 :read data from 7th analysis bin\ kamp, kfreq, 1 ; finction 1 is a stored sine
Page 44-4
The example below shows an example using pvbufread with pvinterp to interpolate between the sound of an oboe and the sound of a clarinet. The value of kinterp returned by a linseg is used to determine the timing of the transitions between the two sounds. The interpolation of frequencies and amplitudes are controlled by the same factor in this example, but for other effects it might be interesting to not have them synchronized in this way. In this example the sound will begin as a clarinet, transform into the oboe and then return again to the clarinet sound. The value of kfreqscale2 is 1.065 because the oboe in this case is a semitone higher in pitch than the clarinet and this brings them approximately to the same pitch. The value of kampscale2 is .75 because the analyzed clarinet was somewhat louder than the analyzed oboe. The setting of these two parameters make the transition quite smooth in this case, but such adjustments are by no means necessary or even advocated.
ktime1 ktime2 kinterp apv 0, p3, 3.5 ; used as index in the oboe.pvoc file line 0, p3, 4.5 ; used as index in the clar.pvoc file line 1, p3*.15, 1, p3*.35, 0, p3*.25, 0, p3*.15, 1, p3*.1, 1 linseg pvbufread ktime1, oboe.pvoc pvinterp ktime2,1,clar.pvoc,1,1.065,1,.75,1-kinterp,1-kinterp
Below is an example using pvbufread with pvcross. In this example the amplitudes used in the resynthesis gradually change from those of the oboe to those of the clarinet. The frequencies, of course, remain those of the clarinet throughout the process since pvcross does not use the frequency data from the file read by pvbufread.
ktime1 ktime2 kcross apv line line expon pvbufread pvcross 0, p3, 3.5 ; used as index in the oboe.pvoc file 0, p3, 4.5 ; used as index in the clar.pvoc file .001, p3, 1 ktime1, oboe.pvoc ktime2, 1, clar.pvoc, 1-kcross, kcross
AUTHOR
Richard Karpen Seattle, Wash 1997
Page 44-5
44.3
ar
pvadd
pvadd ktimpnt, kfmod, ifilcod, ifn, ibins[, ibinoffset, \\ ibinincr, iextractmode, ifreqlim, igatefn]
DESCRIPTION
pvadd reads from a pvoc file and uses the data to perform additive synthesis using an internal array of interpolating oscillators. The user supplies the wave table (usually one period of a sine wave), and can choose which analysis bins will be used in the re-synthesis.
ifilcod integer or character-string denoting a control-file derived from analysis of an audio signal. An integer denotes the suffix of a file pvoc.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not fullpath, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined). pvoc control files contain data organized for fft resynthesis. Memory usage depends on the size of the files involved, which are read and held entirely in memory during computation but are shared by multiple calls (see also lpread). ifn table number of a stored function containing a sine wave ibins number of bins that will be used in the resynthesis (each bin counts as one oscillator in the re-synthesis) ibinoffset (optional) is the first bin used (it is optional and defaults to 0). ibinincr (optional) sets an increment by which pvadd counts up from ibinoffset for ibins components in the re-synthesis (see below for a further explanation). iextractmode (optional) determines if spectral extraction will be carried out and if so whether components that have changes in frequency below ifreqlim or above ifreqlim will be discarded. A value for iextractmode of 1 will cause pvadd to synthesize only those components where the frequency difference between analysis frames is greater than ifreqlim. A value of 2 for iextractmode will cause pvadd to synthesize only those components where the frequency difference between frames is less than ifreqlim. The default values for iextractmode and ifreqlim are 0, in which case a simple resynthesis will be done. See examples below. igatefn (optional) is the number of a stored function which will be applied to the amplitudes of the analysis bins before resynthesis takes place. If igatefn is greater than 0 the amplitudes of each bin will be scaled by igatefn through a simple mapping process. First, the amplitudes of all of the bins in all of the frames in the entire analysis file are compared to determine the maximum amplitude value. This value is then used create normalized amplitudes as indeces into the stored function igatefn. The maximum amplitude will map to the last point in the function. An amplitude of 0 will map to the first point in the function. Values between 0 and 1 will map accordingly to points along the function table.This will be made clearer in the examples below.
PERFORMANCE
ktimpnt and kfmod are used in the same way as in pvoc.
Page 44-6
E X A M PL E S
ktime line 0, p3, p3 asig pvadd ktime, 1, oboe.pvoc, 1, 100, 2
In the above, ibins is 100 and ibinoffset is 2. Using these settings the resynthesis will contain 100 components beginning with bin #2 (bins are counted starting with 0). That is, resynthesis will be done using bins 2-101 inclusive. It is usually a good idea to begin with bin 1 or 2 since the 0th and often 1st bin have data that is neither necessary nor even helpful for creating good clean resynthesis.
ktime line 0, p3, p3 asig pvadd ktime, 1, oboe.pvoc, 1, 100, 2, 2
The above is the same as the previous example with the addition of the value 2 used for the optional ibinincr argument. This result will still result in 100 components in the resynthesis, but pvadd will count through the bins by 2 instead of by 1. It will use bins 2, 4, 6, 8, 10, and so on. For ibins=10, ibinoffset=10, and ibinincr=10, pvadd would use bins 10, 20, 30, 40, up to and including 100. Below is an example using spectral extraction. In this example iextractmode is one and ifreqlim is 9. This will cause pvadd to synthesize only those bins where the frequency deviation, averaged over 6 frames, is greater than 9.
ktime line 0, p3, p3 asig pvadd ktime, 1, "oboe.pvoc", 1, 100, 2, 2, 1, 9
If iextractmode were 2 in the above, then only those bins with an average frequency deviation of less than 9 would be synthesized. If tuned correctly, this technique can be used to separate the pitched parts of the spectrum from the noisy parts. In practice, this depends greatly on the type of sound, the quality of the recording and digitization, and also on the analysis window size and frame increment. Next is an example using amplitude gating. The last 2 in the argument list stands for f2 in the score.
asig pvadd ktime, 1, oboe.pvoc, 1, 100, 2, 2, 0, 0, 2
Suppose the score for the above were to contain:

f2 0 512 7 0 256 1 256 1
Then those bins with amplitudes of 50% of the maximum or greater would be left unchanged, while those with amplitudes less than 50% of the maximum would be scaled down. In this case the lower the amplitude the more severe the scaling down would be. But suppose the score contains:
f2 0 512 5 1 512 .001
In this case, lower amplitudes will be left unchanged and greater ones will be scaled down, turning the sound upside-down in terms of the amplitude spectrum! Functions can be arbitrarily complex. Just remember that the normalized amplitude values of the analysis are themselves the indeces into the function. Finally, both spectral extraction and amplitude gating can be used together. The example below will synthesize only those components that with a frequency deviation of less than 5Hz per frame and it will scale the amplitudes according to F2.
asig pvadd ktime, 1, oboe.pvoc, 1, 100, 1, 1, 2, 5, 2
Page 44-7
USEFUL HINTS:
By using several pvadd units together, one can gradually fade in different parts of the resynthesis, creating various filtering effects. The author uses pvadd to synthesis one bin at a time to have control over each separate component of the re-synthesis. If any combination of ibins, ibinoffset, and ibinincr, creates a situation where pvadd is asked to used a bin number greater than the number of bins in the analysis, it will just use all of the available bins, and give no complaint. So to use every bin just make ibins a big number (i.e. 2000). Expect to have to scale up the amplitudes by factors of 10-100, by the way.
AUTHOR
Richard Karpen Seattle, Wash 1998 (New in Csound version 3.48, additional arguments version 3.56)
Page 44-8
45
S I G N A L G E N E R A T O R S : LPC RESYNTHESIS
lpread, lpreson, lpfreson
lpread ktimpnt, ifilcod[, inpoles[, ifrmrate]] asig, kfrqratio
45.1
krmsr,krmso, kerr,kcps ar ar
lpreson asig lpfreson
DESCRIPTION
These units, used as a read/reson pair, use a control file of time-varying filter coefficients to dynamically modify the spectrum of an audio signal.
ifilcod integer or character-string denoting a control-file (reflection coefficients and four parameter values) derived from n-pole linear predictive spectral analysis of a source audio signal. An integer denotes the suffix of a file lp.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not fullpath, the file is sought first in the current directory, then in that of the environment variable SADIR (if defined). Memory usage depends on the size of the file, which is held entirely in memory during computation but shared by multiple calls (see also adsyn, pvoc). inpoles, ifrmrate (optional) number of poles, and frame rate per second in the lpc analysis. These arguments are required only when the control file does not have a header; they are ignored when a header is detected. The default value for both is zero.
PERFORMANCE
lpread accesses a control file of time-ordered information frames, each containing n-pole filter coefficients derived from linear predictive analysis of a source signal at fixed time intervals (e.g. 1/100 of a second), plus four parameter values: krmsr krmso kerr kcps root-mean-square (rms) of the residual of analysis, rms of the original signal, the normalized error signal, pitch in Hz.
lpread gets its values from the control file according to the input value ktimpnt (in seconds). If ktimpnt proceeds at the analysis rate, time-normal synthesis will result; proceeding at a faster, slower, or variable rate will result in time-warped synthesis. At each k-period, lpread interpolates between adjacent frames to more accurately determine the parameter values (presented as output) and the filter coefficient settings (passed internally to a subsequent lpreson). The error signal kerr (between 0 and 1) derived during predictive analysis reflects the deterministic/random nature of the analyzed source. This will emerge low for pitched (periodic) material and higher for noisy material. The transition from voiced to unvoiced speech, for example, produces an error signal value of about .001. During synthesis, the error signal value can be used to determine the nature of the lpreson driving function: for example, by arbitrating between pitched and non-pitched input, or even by determining a
Signal Generators: LPC Resynthesis
Page 45-1
mix of the two. In normal speech resynthesis, the pitched input to lpreson is a wideband periodic signal or pulse train derived from a unit such as buzz, and the non-pitched source is usually derived from rand. However, any audio signal can be used as the driving function, the only assumption of the analysis being that it has a flat response. lpfreson is a formant shifted lpreson, in which kfrqratio is the (Hz) ratio of shifted to original formant positions. This permits synthesis in which the source object changes its apparent acoustic size. lpfreson with kfrqratio = 1 is equivalent to lpreson. Generally, lpreson provides a means whereby the time-varying content and spectral shaping of a composite audio signal can be controlled by the dynamic spectral content of another. There can be any number of lpread/lpreson (or lpfreson) pairs in an instrument or in an orchestra; they can read from the same or different control files independently.
Page 45-2
45.2
lpslot, lpinterp
lpslot lpinterpol islot islot1, islot2, kmix
DESCRIPTION
Interpolate between two lpc analysis files.
islot number of slot to be selected [0<islot<20] lpslot selects the slot to be use by further lp opcodes. This is the way to load and reference several analysis at the same time. islot1 slot where the first analysis was stored islot2 slot where the second analysis was stored kmix mix value between the two analysis. Should be between 0 and 1. 0 means analysis 1 only. 1 means analysis 2 only. Any value in between will produce interpolation between the filters. lpinterp computes a new set of poles from the interpolation between two analysis. The poles will be stored in the current lpslot and used by the next lpreson opcode.
E X A M PL E
Here is a typical orc using the opcodes:
ipower ifreq asrc ktime init init buzz 50000 ; Define sound generator 440 ipower,ifreq,10,1 0,p3,p3 ; Define time lin 0 ; Read square data poles lpread ktime,square.pol ; Read triangle data poles lpread ktime,triangle.pol 0,p3,1 ; Compute result of mixing 0,1,kmix ; and balance power asrc ares,asrc aout
line lpslot krmsr,krmso,kerr,kcps lpslot 1 krmsr,krmso,kerr,kcps kmix line lpinterp ares lpreson aout balance out
AUTHOR
Mark Resibois Brussels 1996
Page 45-3
Page 45-4
46
SIGNAL GENERATORS: RANDOM (N O I S E ) G E N E R A T O R S

rand, randh, randi
rand randh randi rand randh randi xamp [, iseed[, isize[, ioffset]]] kamp, kcps[, iseed[, isize[, ioffset]]] kamp, kcps[, iseed[, isize[, ioffset]]] xamp [, iseed[, isize[, ioffset]]] xamp, xcps[, iseed[, isize[, ioffset]]] xamp, xcps[, iseed[, isize[, ioffset]]]
46.1
kr kr kr ar ar ar
DESCRIPTION
Output is a controlled random number series between +amp and -amp
iseed (optional) seed value for the recursive pseudo-random formula. A value between 0 and +1 will produce an initial output of kamp * iseed. A negative value will cause seed reinitialization to be skipped. A value greater than 1 will obtain the seed value from the system clock. (New in Csound version 4.10.) The default seed value is .5. isize if zero, a 16 bit number is generated. If non-zero, a 31-bit random number is generated. Default is 0.
PERFORMANCE
koffset (optional) a base value added to the random result. New in Csound version 4.03. The internal pseudo-random formula produces values which are uniformly distributed over the range kamp to -kamp. rand will thus generate uniform white noise with an RMS value of kamp / root 2. The remaining units produce band-limited noise: the cps parameters permit the user to specify that new random numbers are to be generated at a rate less than the sampling or control frequencies. randh will hold each new number for the period of the specified cycle; randi will produce straight-line interpolation between each new number and the next.
E X A M PL E
i1 k1 a1 = randh oscil octpch(p5) ; center pitch, to be modified 1,10 ;10 time/sec by random displacements up to 1 octave 5000, cpsoct(i1+k1), 1
Signal Generators: Random (Noise) Generators
Page 46-1
46.2
ir kr ar ir kr ar ir kr ar ir kr ar ir kr ar ir kr ar ir kr ar ir kr ar ir kr ar ir kr ar ir kr ar
x-class noise generators

linrand linrand linrand trirand trirand trirand exprand exprand exprand bexprnd bexprnd bexprnd cauchy cauchy cauchy cauchy pcauchy pcauchy pcauchy poisson poisson poisson gauss gauss gauss weibull weibull weibull betarand betarand betarand unirand unirand unirand krange krange krange krange krange krange krange krange krange krange krange krange kalpha kalpha kalpha kalpha kalpha kalpha klambda klambda klambda krange krange krange ksigma, ktau ksigma, ktau ksigma, ktau krange, kalpha, kbeta krange, kalpha, kbeta krange, kalpha, kbeta krange krange krange
DESCRIPTION
All of the following opcodes operate in i-, k- and a-rate. linrand krange Linear distribution random number generator. krange is the range of the random numbers (0 krange). Outputs only positive numbers. trirand krange Same as above only outputs both negative and positive numbers. exprand krange Exponential distribution random number generator. krange is the range of the random numbers (0 krange). Outputs only positive numbers. bexprnd krange Same as above, only extends to negative numbers too with an exponential distribution.
The Public Csound Reference Manual Version 4.10 Signal Generators: Random (Noise) Generators Page 46-2
cauchy kalpha -Cauchy distribution random number generator. kalpha controls the spread from zero (big kalpha = big spread). Outputs both positive and negative numbers. pcauchy kalpha Same as above, outputs positive numbers only. poisson klambda Poisson distribution random number generator. klambda is the mean of the distribution. Outputs only positive numbers. gauss krange Gaussian distribution random number generator. krange is the range of the random numbers (-krange 0 krange). Outputs both positive and negative numbers. weibull ksigma, ktau Weibull distribution random number generator. ksigma scales the spread of the distribution and ktau, if greater than one numbers near ksigma are favored, if smaller than one small values are favored and if t equals 1 the distribution is exponential. Outputs only positive numbers. betarand krange, kalpha, kbeta Beta distribution random number generator. krange is the range of the random numbers (0 krange). If kalpha is smaller than one, smaller values favor values near 0. If kbeta is smaller than one, smaller values favor values near krange. If both kalpha and kbeta equal one we have uniform distribution. If both kalpha and kbeta are greater than one we have a sort of Gaussian distribution. Outputs only positive numbers. unirand krange Uniform distribution random number generator. krange is the range of the random numbers (0 krange). For more detailed explanation of these distributions, see: C. Dodge T.A. Jerse 1985. Computer music. Schirmer books. pp.265 286 D. Lorrain. A panoply of stochastic cannons. In C. Roads, ed. 1989. Music machine . Cambridge, Massachusetts: MIT press, pp. 351 379.
E X A M PL E
a1 k1 i1 trirand cauchy betarand 32000 ; Audio noise with triangle distribution 10000 ; Control noise with Cauchy dist. 30000, .5, .5 ; i-time random value, beta dist.
DEPRECATED NAMES
These opcode names originally started with i, k, or a to denote the rate at which the opcode operated. These names are deprecated as of Csound version 3.49. The current form should now be used; the previous form will not work.
AUTHOR
Page 46-3
46.3
ar
pinkish
pinkish xin[, imethod, inumbands, iseed, iskip]
DESCRIPTION
Generates approximate pink noise (-3dB/oct response) by one of two different methods: a multirate noise generator after Moore, coded by Martin Gardner a filter bank designed by Paul Kellet
imethod (optional) selects filter method: 0 = Gardner method (default). 1 = Kellet filter bank. 2 = A somewhat faster filter bank by Kellet, with less accurate response. inumbands (optional) only effective with Gardner method. The number of noise bands to generate. Maximum is 32, minimum is 4. Higher levels give smoother spectrum, but above 20 bands there will be almost DC-like slow fluctuations. Default value is 20. iseed (optional) only effective with Gardner method. If non-zero, seeds the random generator. If zero, the generator will be seeded from current time. Default is 0. iskip (optional) if non-zero, skip (re)initialization of internal state (useful for tied notes). Default is 0.
PERFORMANCE
xin for Gardner method: k- or a-rate amplitude. For Kellet filters: normally a-rate uniform random noise from rand (31-bit) or unirand, but can be any a-rate signal. The output peak value varies widely (15%) even over long runs, and will usually be well below the input amplitude. Peak values may also occasionally overshoot input amplitude or noise. pinkish attempts to generate pink noise (i.e., noise with equal energy in each octave), by one of two different methods. The first method, by Moore & Gardner, adds several (up to 32) signals of white noise, generated at octave rates (sr, sr/2, sr/4 etc). It obtains pseudo-random values from an internal 32-bit generator. This random generator is local to each opcode instance and seedable (similar to rand). The second method is a lowpass filter with a response approximating -3dB/oct. If the input is uniform white noise, it outputs pink noise. Any signal may be used as input for this method. The high quality filter is slower, but has less ripple and a slightly wider operating frequency range than less computationally intense versions. With the Kellet filters, seeding is not used. The Gardner method output has some frequency response anomalies in the low-mid and high-mid frequency ranges. More low-frequency energy can be generated by increasing the number of bands. It is also a bit faster. The refined Kellet filter has very smooth spectrum, but a more limited effective range. The level increases slightly at the high end of the spectrum.
Page 46-4
E X A M PL E
Kellet-filtered noise for a tied note (iskip is non-zero).
instr 1 unirand = pinkish out endin
awhite awhite apink
2.0 awhite - 1.0 awhite, 1, 0, 0, 1 apink * 30000
; Normalize to +/-1.0
AUTHORS
Phil Burke John ffitch University of Bath/Codemist Ltd. Bath, UK May, 2000 (New in Csound version 4.06)
Page 46-5
46.4
ar
noise
noise xamp, kbeta
DESCRIPTION
A white noise generator with an IIR lowpass filter.
PERFORMANCE
xamp amplitude of final output kbeta beta of the lowpass filter. Should be in the range of 0 to 1. The filter equation is: y_n = sqrt(1-beta^2) * x_n + beta Y_(n-1) where x_n is white noise.
AUTHOR
John ffitch University of Bath, Codemist. Ltd. Bath, UK December, 2000 New in Csound version 4.10
Page 46-6
47
FUNCTION TABLE CONTROL: TABLE QUERIES

ftlen, ftlptim, ftsr, nsamp
ftlen(x) ftlen ftlptim(x) ftlptim ftsr(x) ftsr nsamp(x) nsamp (init-rate (init-rate (init-rate (init-rate args args args args only) only) only) only)
47.1
DESCRIPTION
Where the argument within the parentheses may be an expression. These value converters return information about a stored function table. The result can be a term in a further expression.
PERFORMANCE
ftlen(x) returns the size (number of points, excluding guard point) of stored function table number x. While most units referencing a stored table will automatically take its size into account (so tables can be of arbitrary length), this function reports the actual size, if that is needed. Note that ftlen will always return a power-of-2 value, i.e. the function table guard point (see f Statement) is not included. As of Csound version 3.53, ftlen works with deferred function tables (see GEN01). ftlptim(x) returns the loop segment start-time (in seconds) of stored function table number x. This reports the duration of the direct recorded attack and decay parts of a sound sample, prior to its looped segment. Returns zero (and a warning message) if the sample does not contain loop points. ftsr(x) returns the sampling-rate of a GEN01 or GEN22 generated table. The samplingrate is determined from the header of the original file. If the original file has no header, or the table was not created by these two GENs ftsr returns 0. New in Csound version 3.49. nsamp(x) returns the number of samples loaded into stored function table number x by GEN01 or GEN23. This is useful when a sample is shorter than the power-of-two function table that holds it. New in Csound version 3.49.
AUTHORS
Barry Vercoe MIT Cambridge, Massachusetts 1997 Gabriel Maldonado (ftsr, nsamp) Italy October, 1998
Function Table Control: Table Queries
Page 47-1
47.2
ir kr
tableng
tableng tableng ifn kfn
DESCRIPTION
Interrogates a function table for length.
47.2.1 47.2.2
I N I T I A L I ZA T I O N ifn Table number to be interrogated PERFORMANCE kfn Table number to be interrogated

tableng returns the length of the specified table. This will be a power of two number in most circumstances. It will not show whether a table has a guardpoint or not. It seems this information is not available in the tables data structure. If the specified table is not found, then 0 will be returned. Likely to be useful for setting up code for table manipulation operations, such as tablemix and tablecopy.
N A M E C HA N G ES
As of Csound version 3.52, the name of the opcode itablegpw has been changed to tableng.
AUTHOR
Page 47-2
48
FUNCTION TABLE CONTROL: TABLE SELECTION

tablekt, tableikt
tablekt tablekt tableikt tableikt kndx, xndx, kndx, xndx, kfn[, kfn[, kfn[, kfn[, ixmode[, ixmode[, ixmode[, ixmode[, ixoff[, ixoff[, ixoff[, ixoff[, iwrap]]] iwrap]]] iwrap]]] iwrap]]]
48.1
kr ar kr ar
DESCRIPTION
k-rate control over table numbers. The standard Csound opcodes table and tablei, when producing a k- or a-rate result, can only use an init-time variable to select the table number. tablekt and tableikt accept krate control as well as i-time. In all other respects they are similar to the original opcodes.
indx Index into table, either a positive number range ifn Table number. Must be >= 1. Floats are rounded down to an integer. If a table number does not point to a valid table, or the table has not yet been loaded (GEN01) then an error will result and the instrument will be de-activated. ixmode if 0, xndx and ixoff ranges match the length of the table. if non-zero xndx and ixoff have a 0 to 1 range. Default is 0 ixoff if 0, total index is controlled directly by xndx, i.e. the indexing starts from the start of the table. If non-zero, start indexing from somewhere else in the table. Value must be positive and less than the table length (ixmode = 0) or less than 1 (ixmode !=0). Default is 0. iwrap if iwrap = 0, Limit mode: when total index is below 0, then final index is 0.Total index above table length results in a final index of the table length high out of range total indexes stick at the upper limit of the table. If iwrap !=0, Wrap mode: total index is wrapped modulo the table length so that all total indexes map into the table. For instance, in a table of length 8, xndx = 5 and ixoff = 6 gives a total index of 11, which wraps to a final index of 3. Default is 0.
PERFORMANCE
kndx Index into table, either a positive number range xndx matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0) kfn Table number. Must be >= 1. Floats are rounded down to an integer. If a table number does not point to a valid table, or the table has not yet been loaded (GEN01) then an error will result and the instrument will be de-activated.
Function Table Control: Table Selection
Page 48-1
AUTHOR
Robin Whittle Australia 1997
Page 48-2
49
FUNCTION TABLE CONTROL: R E A D /W R I T E O P E R A T I O N S

tableiw, tablew, tablewkt
tableiw tablew tablew tablewkt tablewkt isig, ksig, asig, ksig, asig, indx, kndx, andx, kndx, andx, ifn[, ifn[, ifn[, kfn[, kfn[, ixmode[, ixmode[, ixmode[, ixmode[, ixmode[, ixoff[, ixoff[, ixoff[, ixoff[, ixoff[, iwgmode]]] iwgmode]]] iwgmode]]] iwgmode]]] iwgmode]]]
49.1
DESCRIPTION
These opcodes operate on existing function tables, changing their contents. tableiw is used when all inputs are init time variables or constants and you only want to run it at the initialization of the instrument. tablew is for writing at k- or at a-rates, with the table number being specified at init time. tablewkt is the same, but uses a k-rate variable for selecting the table number. The valid combinations of variable types are shown by the first letter of the variable names.
isig, ksig, asig- The value to be written into the table. indx, kndx, andx Index into table, either a positive number range matching the table length (ixmode = 0) or a 0 to 1 range (ixmode != 0) ifn, kfn Table number. Must be = 1. Floats are rounded down to an integer. If a table number does not point to a valid table, or the table has not yet been loaded (GEN01) then an error will result and the instrument will be de-activated. ixmode Default is 0. 0 : xndx and ixoff ranges match the length of the table. !=0 : xndx and ixoff have a 0 to 1 range. ixoff Default is 0. 0: Total index is controlled directly by xndx. i.e. the indexing starts from the start of the table. !=0: Start indexing from somewhere else in the table. Value must be positive and less than the table length (ixmode = 0) or less than 1 (ixmode !=0). iwgmode Default is 0. 0: Limit mode 1: Wrap mode 2: Guardpoint mode.
PERFORMANCE
Limit mode (0) Limit the total index (ndx + ixoff) to between 0 and the guard point. For a table of length 5, this means that locations 0 to 3 and location 4 (the guard point) can be written. A negative total index writes to location 0. Total indexes 4 write to location 4.
Function Table Control: Read/Write Operations
Page 49-1
Wrap mode (1) Wrap total index value into locations 0 to E, where E is one less than either the table length or the factor of 2 number which is one less than the table length. For example, wrap into a 0 to 3 range so that total index 6 writes to location 2. Guardpoint mode (2) The guardpoint is written at the same time as location 0 is written with the same value. This facilitates writing to tables which are intended to be read with interpolation for producing smooth cyclic waveforms. In addition, before it is used, the total index is incremented by half the range between one location and the next, before being rounded down to the integer address of a table location. Normally (igwmode = 0 or 1) for a table of length 5 which has locations 0 to 3 as the main table and location 4 as the guard point, a total index in the range of 0 to 0.999 will write to location 0. (0.999 means just less than 1.0.) 1.0 to 1.999 will write to location 1, etc. A similar pattern holds for all total indexes 0 to 4.999 (igwmode = 0) or to 3.999 (igwmode = 1). igwmode = 0 enables locations 0 to 4 to be written with the guardpoint (4) being written with a potentially different value from location 0. With a table of length 5 and the iwgmode = 2, then when the total index is in the range 0 to 0.499, it will write to locations 0 and 4. Range 0.5 to 1.499 will write to location 1 etc. 3.5 to 4.0 will also write to locations 0 and 4. This way, the writing operation most closely approximates the results of interpolated reading. Guard point mode should only be used with tables that have a guardpoint. Guardpoint mode is accomplished by adding 0.5 to the total index, rounding to the next lowest integer, wrapping it modulo the factor of two which is one less than the table length, writing the table (locations 0 to 3 in our example) and then writing to the guard point if index == 0. tablew has no output value. The last three parameters are optional and have default values of 0. Caution with k-rate table numbers : The following notes also apply to the tablekt and tableikt opcodes which can now have their table number changed at k-rate. At k-rate or a-rate, if a table number of < 1 is given, or the table number points to a nonexistent table, or to one which has a length of 0 (it is to be loaded from a file later) then an error will result and the instrument will be deactivated. kfn and afn must be initialized at the appropriate rate using init. Attempting to load an i-rate value into kfn or afn will result in an error.
C HA N GE D N A M E
As of Csound version 3.52, the opcode name itablew is changed to tableiw.
AUTHOR
Page 49-2
49.2
tablegpw, tablemix, tablecopy, tableigpw, tableimix, tableicopy

tablegpw tablemix tablecopy tableigpw tableimix tableicopy kfn kdft, kdoff, klen, ks1ft, ks1off, ks1g, ks2ft, \\ ks2off, ks2g kdft, ksft ifn idft, idoff, ilen, is1ft, is1off, is1g, is2ft, \\ is2off, is2g idft, isft
DESCRIPTION
These opcodes allow tables to be copied and mixed.
ifn Function table number
PERFORMANCE
kfn Function table number kdft Destination function table number kdoff Offset to start writing from. Can be negative. kdft Number of destination function table. ksft Number of source function table. klen Number of write operations to perform. Negative means work backwards. ks1ft, ks2ft Source function tables. These can be the same as the destination table, if care is exercised about direction of copying data. ks1off, ks2off Offsets to start reading from in source tables. ks1g, ks2g Gains to apply when reading from the source tables. The results are added and the sum is written to the destination table. tablgpw For writing the tables guard point, with the value which is in location 0. Does nothing if table does not exist. Likely to be useful after manipulating a table with tablemix or tablecopy. tablemix This opcode mixes from two tables, with separate gains into the destination table. Writing is done for klen locations, usually stepping forward through the table if klen is positive. If it is negative, then the writing and reading order is backwards towards lower indexes in the tables. This bi-directional option makes it easy to shift the contents of a table sideways by reading from it and writing back to it with a different offset. If klen is 0, no writing occurs. Note that the internal integer value of klen is derived from the ANSI C floor() function which returns the next most negative integer. Hence a fractional negative klen value of -2.3 would create an internal length of 3, and cause the copying to start from the offset locations and proceed for two locations to the left. The total index for table reading and writing is calculated from the starting offset for each table, plus the index value, which starts at 0 and then increments (or decrements) by 1 as mixing proceeds.
The Public Csound Reference Manual Version 4.10 Function Table Control: Read/Write Operations Page 49-3
These total indexes can potentially be very large, since there is no restriction on the offset or the klen. However each total index for each table is ANDed with a length mask (such as 0000 0111 for a table of length 8) to form a final index which is actually used for reading or writing. So no reading or writing can occur outside the tables. This is the same as wrap mode in table read and write. These opcodes do not read or write the guardpoint. If a table has been rewritten with one of these, then if it has a guardpoint which is supposed to contain the same value as the location 0, then call tablegpw afterwards. The indexes and offsets are all in table steps they are not normalized to 0 1. So for a table of length 256, klen should be set to 256 if all the table was to be read or written. The tables do not need to be the same length wrapping occurs individually for each table. tablecopy Simple, fast table copy opcodes. Takes the table length from the destination table, and reads from the start of the source table. For speed reasons, does not check the source length just copies regardless in wrap mode. This may read through the source table several times. A source table with length 1 will cause all values in the destination table to be written to its value. tablecopy cannot read or write the guardpoint. To read it use table, with ndx = the table length. Likewise use table write to write it. To write the guardpoint to the value in location 0, use tablegpw. This is primarily to change function tables quickly in a real-time situation.
N A M E C HA N G ES
As of Csound version 3.52, the names of the opcodes itablegpw, itablemix, and itablecopy, have been changed to tableigpw, tableimix, and tableicopy, respectively.
AUTHOR
Page 49-4
49.3
ar kstart
tablera, tablewa
tablera tablewa kfn, kstart, koff kfn, asig, koff
DESCRIPTION
These opcodes read and write tables in sequential locations to and from an a-rate variable. Some thought is required before using them. They have at least two major, and quite different, applications which are discussed below.
ar a-rate destination for reading ksmps values from a table. kfn i- or k-rate number of the table to read or write. kstart Where in table to read or write. asig a-rate signal to read from when writing to the table. koff i- or k-rate offset into table. Range unlimited see explanation at end of this section.
PERFORMANCE
In one application, these are intended to be used in pairs, or with several tablera opcodes before a tablewa all sharing the same kstart variable. These read from and write to sequential locations in a table at audio rates, with ksmps floats being written and read each cycle. tablera starts reading from location kstart. tablewa starts writing to location kstart, and then writes to kstart with the number of the location one more than the one it last wrote. (Note that for tablewa, kstart is both an input and output variable.) If the writing index reaches the end of the table, then no further writing occurs and zero is written to kstart. For instance, if the tables length was 16 (locations 0 to 15), and ksmps was 5. Then the following steps would occur with repetitive runs of the tablewa opcode, assuming that kstart started at 0. Run no. Initial kstart 1 0 2 5 3 10 4 15 Final kstart 5 10 15 0 locations written 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
This is to facilitate processing table data using standard a-rate orchestra code between the tablera and tablewa opcodes. They allow all Csound k-rate operators to be used (with caution) on a-rate variables something that would only be possible otherwise by ksmps = 1, downsamp and upsamp.
Page 49-5
Several cautions: The k-rate code in the processing loop is really running at a-rate, so time dependant functions like port and oscil work faster than normal their code is expecting to be running at k-rate. This system will produce undesirable results unless the ksmps fits within the table length. For instance a table of length 16 will accommodate 1 to 16 samples, so this example will work with ksmps = 1 to 16. Both these opcodes generate an error and deactivate the instrument if a table with length < ksmps is selected. Likewise an error occurs if kstart is below 0 or greater than the highest entry in the table if kstart = table length. kstart is intended to contain integer values between 0 and (table length 1). Fractional values above this should not affect operation but do not achieve anything useful. These opcodes are not interpolating, and the kstart and koff parameters always have a range of 0 to (table length 1) not 0 to 1 as is available in other table read/write opcodes. koff can be outside this range but it is wrapped around by the final AND operation. These opcodes are permanently in wrap mode. When koff is 0, no wrapping needs to occur, since the kstart++ index will always be within the tables normal range. koff != 0 can lead to wrapping. The offset does not affect the number of read/write cycles performed, or the value written to kstart by tablewa. These opcodes cannot read or write the guardpoint. Use tablegpw to write the guardpoint after manipulations have been done with tablewa.
E X A M PL E S
kstart lab1: atemp atemp = 0 ; Read 5 values from table into an ; a-rate variable.
tablera ktabsource, kstart, 0 = log(atemp)
; Process the values using a-rate ; code. ; Write it back to the table ; Loop until all table locations ; have been processed.
kstart tablewa ktabdest, atemp, 0 if ktemp 0 goto lab1
The above example shows a processing loop, which runs every k-cycle, reading each location in the table ktabsource, and writing the log of those values into the same locations of table ktabdest. This enables whole tables, parts of tables (with offsets and different control loops) and data from several tables at once to be manipulated with a-rate code and written back to another (or to the same) table. This is a bit of a fudge, but it is faster than doing it with krate table read and write code. Another application is:
kzero = 0 kloop = 0 kzero tablewa 23, asignal, 0 lab1: ktemp table kloop, 23 ; ksmps a-rate samples written ; into locations 0 to (ksmps -1) of table 23.
; Start a loop which runs ksmps times, ; in which each cycle processes one of [ Some code to manipulate ] ; table 23s values with k-rate orchestra [ the value of ktemp. ] ; code.
Page 49-6
tablew ktemp, kloop, 23 ; Write the processed value to the table. kloop = kloop + 1 if kloop < ksmps goto lab1 asignal tablera 23, 0, 0 ; Increment the kloop, which is both the ; pointer into the table and the loop ; counter. Keep looping until all values ; in the table have been processed. ; Copy the table contents back ; to an a-rate variable.
koff This is an offset which is added to the sum of kstart and the internal index variable which steps through the table. The result is then ANDed with the lengthmask (000 0111 for a table of length 8 or 9 with guardpoint) and that final index is used to read or write to the table. koff can be any value. It is converted into a long using the ANSI floor() function so that -4.3 becomes -5. This is what we would want when using offsets which range above and below zero. Ideally this would be an optional variable, defaulting to 0, however with the existing Csound orchestra read code, such default parameters must be init time only. We want krate here, so we cannot have a default.
Page 49-7
Page 49-8
50
SIGNAL MODIFIERS: STANDARD FILTERS

port, portk, tone, tonek, atone, atonek, reson, resonk, areson, aresonk
port portk tonek atonek resonk aresonk tone atone reson areson ksig, ksig, ksig, ksig, ksig, ksig, asig, asig, asig, asig, ihtim[, isig] khtim[, isig] khp[, iskip] khp[, iskip] kcf, kbw[, iscl, kcf, kbw[, iscl, khp[, iskip] khp[, iskip] kcf, kbw[, iscl, kcf, kbw[, iscl,
50.1
kr kr kr kr kr kr ar ar ar ar
iskip] iskip] iskip] iskip]
DESCRIPTION
A control or audio signal is modified by a low- or band-pass recursive filter with variable frequency response.
isig initial (i.e. previous) value for internal feedback. The default value is 0. iskip initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. iscl coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment ( see balance). The default value is 0.
PERFORMANCE
port applies portamento to a step-valued control signal. At each new step value, ksig is low-pass filtered to move towards that value at a rate determined by ihtim. ihtim is the half-time of the function (in seconds), during which the curve will traverse half the distance towards the new value, then half as much again, etc., theoretically never reaching its asymptote. With portk, the half-time can be varied at the control rate. tone implements a first-order recursive low-pass filter in which the variable khp (in Hz) determines the response curves half-power point. Half power is defined as peak power / root 2.
Signal Modifiers: Standard Filters
Page 50-1
reson is a second-order filter in which kcf controls the center frequency, or frequency position of the peak response, and kbw controls its bandwidth (the frequency difference between the upper and lower half-power points). atone, areson are filters whose transfer functions are the complements of tone and reson. atone is thus a form of high-pass filter and areson a notch filter whose transfer functions represent the filtered out aspects of their complements. Note, however, that power scaling is not normalized in atone, areson, but remains the true complement of the corresponding unit. Thus an audio signal, filtered by parallel matching reson and areson units, would under addition simply reconstruct the original spectrum. This property is particularly useful for controlled mixing of different sources ( see lpreson). Complex response curves such as those with multiple peaks can be obtained by using a bank of suitable filters in series. (The resultant response is the product of the component responses.) In such cases, the combined attenuation may result in a serious loss of signal power, but this can be regained by the use of balance.
Page 50-2
50.2
ar ar ar
tonex, atonex, resonx

tonex atonex resonx asig, khp[, inumlayer, iskip] asig, khp[, inumlayer, iskip] asig, kcf, kbw[, inumlayer, iscl, iskip]
DESCRIPTION
tonex, atonex and resonx are equivalent to filters consisting of more layers of tone, atone and reson, with the same arguments, serially connected. Using a stack of a larger number of filters allows a sharper cutoff. They are faster than using a larger number instances in a Csound orchestra of the old opcodes, because only one initialization and k cycle are needed at time, and the audio loop falls entirely inside the cache memory of processor.
inumlayer number of elements in the filter stack.. Default value is 4. iskip initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. iscl coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment ( see balance). The default value is 0.
PERFORMANCE
asig input signal khp the response curves half-power point. Half power is defined as peak power / root 2. kcf the center frequency of the filter, or frequency position of the peak response. kbw bandwidth of the filter (the Hz difference between the upper and lower half-power points)
AUTHOR
Gabriel Maldonado (adapted by John ffitch) Italy New in Csound version 3.49
Page 50-3
50.3
ar ar
resonr, resonz
resonr resonz asig, kcf, kbw[,iscl, iskip] asig, kcf, kbw[,iscl, iskip]
DESCRIPTION
Implementations of a second-order, two-pole two-zero bandpass filter with variable frequency response.
The optional initialization variables for resonr and resonz are identical to the i-time variables for reson. iskip initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. iscl coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise. A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0.
PERFORMANCE
resonr and resonz are variations of the classic two-pole bandpass resonator (reson). Both filters have two zeroes in their transfer functions, in addition to the two poles. resonz has its zeroes located at z = 1 and z = -1. resonr has its zeroes located at +sqrt(R) and -sqrt(R), where R is the radius of the poles in the complex z-plane. The addition of zeroes to resonr and resonz results in the improved selectivity of the magnitude response of these filters at cutoff frequencies close to 0, at the expense of less selectivity of frequencies above the cutoff peak. resonr and resonz are very close to constant-gain as the center frequency is swept, resulting in a more efficient control of the magnitude response than with traditional twopole resonators such as reson. resonr and resonz produce a sound that is considerably different from reson, especially for lower center frequencies; trial and error is the best way of determining which resonator is best suited for a particular application. asig input signal to be filtered kcf cutoff or resonant frequency of the filter, measured in Hz kbw bandwidth of the filter (the Hz difference between the upper and lower half-power points)
Page 50-4
E X A M PL E
; ; ; ; Orchestra file for resonant filter sweep of a sawtooth-like waveform. The outputs of reson, resonr, and resonz are scaled by coefficients specified in the score, so that each filter can be heard on its own from the same instrument. = = = = 44100 4410 10 1 instr 1 idur ibegfreq iendfreq ibw ifreq iamp ires iresr iresz = = = = = = = = = p3 p4 p5 p6 p7 p8 p9 p10 p11 ; ; ; ; ; ; ; ; beginning of sweep frequency ending of sweep frequency bandwidth of filters in Hz frequency of gbuzz that is to be filtered amplitude to scale output by coefficient to scale amount of reson in output coefficient to scale amount of resonr in output coefficient to scale amount of resonz in output
sr kr ksmps nchnls
; Frequency envelope for reson cutoff kfreq ibegfreq, idur * .5, iendfreq, idur * .5, ibegfreq linseg ; Amplitude envelope to prevent clicking kenv linseg 0, .1, iamp, idur - .2, iamp, .1, 0 ; Number of harmonics for gbuzz scaled to avoid aliasing iharms = (sr*.4)/ifreq asig ain ares aresr aresz gbuzz = reson resonr resonz out endin ; Score file for above f1 0 8192 9 1 1 .25 i1 i1 i1 i1 i1 i1 e 0 10 20 30 40 50 10 10 10 10 10 10 1 1 1 1 1 1 3000 200 100 4000 1 0 0 3000 200 100 4000 0 1 0 3000 200 100 4000 0 0 1 3000 50 200 8000 1 0 0 3000 50 200 8000 0 1 0 3000 50 200 8000 0 0 1 ; cosine table for gbuzz generator ; ; ; ; ; ; reson resonr resonz reson resonr resonz output output output output output output with with with with with with bw bw bw bw bw bw = = = = = = 200 200 200 50 50 50 1, ifreq, iharms, 1, .9, 1 kenv * asig ain, kfreq, ibw, 1 ain, kfreq, ibw, 1 ain, kfreq, ibw, 1 ; Sawtooth waveform ; output scaled by amp ; envelope
ares * ires + aresr * iresr + aresz * iresz
TECHNICAL HISTORY
resonr and resonz were originally described in an article by Julius O. Smith and James B. Angell [1]. Smith and Angell recommended the resonz form (zeros at +1 and -1) when computational efficiency was the main concern, as it has one less multiply per sample, while resonr (zeroes at + and the square root of the pole radius R) was recommended for situations when a perfectly constant-gain center peak was required. Ken Steiglitz, in a later article [2], demonstrated that resonz had constant gain at the true peak of the filter, as opposed to resonr, which displayed constant gain at the pole angle. Steiglitz also recommended resonz for its sharper notches in the gain curve at zero and Nyquist frequency. Steiglitzs recent book [3] features a thorough technical discussion of reson and resonz, while Dodge and Jerses textbook [4] illustrates the differences in the response curves of reson and resonz.
Page 50-5
REFERENCES
1. 1. Smith, Julius O. and Angell, James B., A Constant-Gain Resonator Tuned by a Single Coefficient, Computer Music Journal, vol. 6, no. 4, pp. 36-39, Winter 1982. 2. 2. Steiglitz, Ken, A Note on Constant-Gain Digital Resonators, Computer Music Journal, vol. 18, no. 4, pp. 8-10, Winter 1994. 3. 3. Ken Steiglitz, A Digital Signal Processing Primer, with Applications to Digital Audio and Computer Music. Addison-Wesley Publishing Company, Menlo Park, CA, 1996. 4. 4. Dodge, Charles and Jerse, Thomas A., Computer Music: Synthesis, Composition, and Performance. New York: Schirmer Books, 1997, 2nd edition, pp. 211-214.
AUTHOR
Sean Costello Seattle, Washington 1999 New in Csound version 3.55
Page 50-6
50.4
ar
resony
resony asig, kbf, kbw, inum, ksep[, isepmode, iscl, iskip]
DESCRIPTION
A bank of second-order bandpass filters, connected in parallel.
inum number of filters isepmode if isepmode = 0, the separation of center frequencies of each filter is generated logarithmically (using octave as unit of measure). If isepmode != 0, the separation of center frequencies of each filter is generated linearly (using Hertz). Default value is 0. iscl coded scaling factor for resonators. A value of 1 signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A value of 2 raises the response factor so that its overall RMS value equals 1. (This intended equalization of input and output power assumes all frequencies are physically present; hence it is most applicable to white noise.) A zero value signifies no scaling of the signal, leaving that to some later adjustment (e.g. balance). The default value is 0. iskip initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0.
PERFORMANCE
asig audio input signal kbf base frequency, i.e. center frequency of lowest filter in Hz kbw bandwidth in Hz ksep separation of the center frequency of filters in octaves resony is a bank of second-order bandpass filters, with k-rate variant frequency separation, base frequency and bandwidth, connected in parallel (i.e. the resulting signal is a mix of the output of each filter). The center frequency of each filter depends of kbf and ksep variables. The maximum number of filters is set to 100.
E X A M PL E
In this example the global variable gk1 modifies kbf, gk2 modifies kbw, gk3 inum, gk4 ksep, and gk5 the main volume.
a1 a2 instr soundin resony out endin 1 "myfile.aif" a1, gk1, gk2, i(gk3), gk4, 2 a2 * gk5
AUTHOR
Gabriel Maldonado Italy 1999 New in Csound version 3.56
Page 50-7
50.5
ar ar
lowres, lowresx
lowres lowresx asig, kcutoff, kresonance [,iskip] asig, kcutoff, kresonance [, inumlayer, iskip]
DESCRIPTION
lowres is a resonant lowpass filter. lowresx is equivalent to more layer of lowres, with the same arguments, serially connected.
inumlayer number of elements in a lowresx stack. Default value is 4. There is no maximum. iskip initial disposition of internal data space. A zero value will clear the space; a nonzero value will allow previous information to remain. The default value is 0.
PERFORMANCE
asig input signal kcutoff filter cutoff frequency point kresonance resonance amount lowres is a resonant lowpass filter derived from a Hans Mikelson orchestra. This implementation is much faster than implementing it in Csound language, and it allows kr lower than sr. kcutoff is not in Hz and kresonance is not in dB, so experiment for the finding best results. lowresx is equivalent to more layer of lowres, with the same arguments, serially connected. Using a stack of a larger number of filters allows a sharper cutoff. This is faster than using a larger number of instances of lowres in a Csound orchestra, because only one initialization and k cycle are needed at time, and the audio loop falls entirely inside the cache memory of processor. Based on an orchestra by Hans Mikelson.
AUTHOR
Gabriel Maldonado (adapted by John ffitch) Italy New in Csound version 3.49
Page 50-8
50.6
ar
vlowres
vlowres asig, kfco, kres, iord, ksep
DESCRIPTION
A bank of filters in which the cutoff frequency can be separated under user control
iord total number of filters (1 to 10)
PERFORMANCE
asig input signal kfco frequency cutoff (not in Hz) ksep frequency cutoff separation for each filter vlowres (variable resonant lowpass filter) allows a variable response curve in resonant filters. It can be thought of as a bank of lowpass resonant filters, each with the same resonance, serially connected. The frequency cutoff of each filter can vary with the kcfo and ksep parameters.
AUTHOR
Page 50-9
50.7
ar
lowpass2
lowpass2 asig, kcf, kq[, iskip]
DESCRIPTION
Implementation of resonant second-order lowpass filter.
iskip initial disposition of internal data space. A zero value will clear the space; a nonzero value will allow previous information to remain. The default value is 0
PERFORMANCE
asig input signal to be filtered kcf cutoff or resonant frequency of the filter, measured in Hz kq Q of the filter, defined, for bandpass filters, as bandwidth/cutoff. kq should be between 1 and 500 lowpass2 is a second order IIR lowpass filter, with k-rate controls for cutoff frequency (kcf) and Q (kq). As kq is increased, a resonant peak forms around the cutoff frequency, transforming the lowpass filter response into a response that is similar to a bandpass filter, but with more low frequency energy. This corresponds to an increase in the magnitude and sharpness of the resonant peak. For high values of kq, a scaling function such as balance may be required. In practice, this allows for the simulation of the voltage-controlled filters of analog synthesizers, or for the creation of a pitch of constant amplitude while filtering white noise.
E X A M PL E
; Orchestra file for resonant filter sweep of a sawtooth-like waveform. = 44100 sr = 2205 kr ksmps = 20 nchnls = 1 instr 1 idur ifreq iamp iharms = = = = p3 p4 p5 * .5 (sr*.4) / ifreq 1, ifreq, iharms, 1, .9, 1
; Sawtooth-like waveform asig gbuzz
; Envelope to control filter cutoff kfreq linseg 1, idur * 0.5, 5000, idur * 0.5, 1 afilt lowpass2 asig, kfreq, 30
Page 50-10
; Simple amplitude envelope kenv 0, .1, iamp, idur -.2, iamp, .1, 0 linseg asig * kenv out endin ; Score file for above f1 0 8192 9 1 1 .25 i1 0 5 100 1000 i1 5 5 200 1000 e
AUTHOR
Sean Costello Seattle, Washington August, 1999 New in Csound version 4.0
Page 50-11
50.8
ar ar ar
biquad, rezzy, moogvcf

biquad rezzy moogvcf asig, kb0, kb1, kb2, ka0, ka1, ka2[, iskip] asig, xfco, xres[, imode] asig, xfco, xres[, iscale]
DESCRIPTION
Implementation of a sweepable general purpose filter and two sweepable, resonant lowpass filters.
iskip (optional) if non-zero, initialization will be skipped. Default value 0. (New in Csound version 3.50) imode (optional) if zero rezzy is low-pass, if nonzero, high-pass. Default value is 0. (New in Csound version 3.50) iscale (optional) internal scaling factor. Use if asig is not in the range +/-1. Input is first divided by iscale, then output is multiplied iscale. Default value is 1. (New in Csound version 3.50)
PERFORMANCE
asig input signal xfco filter cut-off frequency in Hz. As of version 3.50, may i-,k-, or a-rate. xres amount of resonance. For rezzy, values of 1 to 100 are typical. Resonance should be one or greater. For moogvcf, self-oscillation occurs when xres is approximately one. As of version 3.50, may i-,k-, or a-rate. biquad is a general purpose biquadratic digital filter of the form: a0*y(n) + a1*y[n-1] + a2*y[n-2] = b0*x[n] + b1*x[n-1] + b2*x[n-2] This filter has the following frequency response: B(Z) b0 + b1*Z-1 + b2*Z-2 H(Z) = ---- = -----------------A(Z) a0 + a1*Z-1 + a2*Z-2 This type of filter is often encountered in digital signal processing literature. It allows six user-defined k-rate coefficients. rezzy is a resonant low-pass filter created empirically by Hans Mikelson. moogvcf is a digital emulation of the Moog diode ladder filter configuration. This emulation is based loosely on the paper Analyzing the Moog VCF with Considerations for Digital Implementation by Stilson and Smith (CCRMA). This version was originally coded in Csound by Josep Comajuncosas. Some modifications and conversion to C were done by Hans Mikelson. Note: This filter requires that the input signal be normalized to one.
Page 50-12
E X A M PL E S
;biquad example biquad kfcon = *3.14159265*kfco/sr kalpha = -2*krez*cos(kfcon)*cos(kfcon)+krez*krez*cos(2*kfcon) kbeta = *krez*sin(2*kfcon)-2*krez*cos(kfcon)*sin(kfcon) kgama = +cos(kfcon) km1 = *kgama+kbeta*sin(kfcon) km2 = *kgama-kbeta*sin(kfcon) kden = (km1*km1+km2*km2) kb0 = .5*(kalpha*kalpha+kbeta*kbeta)/kden kb1 = kb0 kb2 = 0 ka0 = 1 ka1 = -2*krez*cos(kfcon) ka2 = krez*krez ayn biquad axn, kb0, kb1, kb2, ka0, ka1, ka2 ayn*iamp/2, ayn*iamp/2 outs ; Sta i14 8.0 i14 + Dur 1.0 1.0 Amp 20000 20000 Pitch Fco 6.00 1000 6.03 2000 Rez .8 .95
;rezzy example rezzy kfco 100+.01*ifco, .2*idur, ifco+100, .5*idur, ifco*.1+100, expseg .3*idur, .001*ifco+100 apulse1 1,ifqc, sr/2/ifqc, 1 ; Avoid aliasing buzz asaw integ apulse1 axn asaw-.5 = ayn axn, kfco, krez rezzy ayn*iamp, ayn*iamp outs ; i10 i10 Sta 0.0 + Dur 1.0 1.0 Amp 20000 20000 Pitch 6.00 6.03 Fco 1000 2000 Rez 2 10
;moogvcf example moogvcf apulse1 1,ifqc, sr/2/ifqc, 1 ; Avoid aliasing buzz asaw integ apulse1 ax = asaw-.5 ayn ax, kfco, krez moogvcf ayn*iamp, ayn*iamp outs ; i11 i11 Sta 4.0 + Dur 1.0 1.0 Amp 20000 20000 Pitch 6.00 6.03 Fco 1000 2000 Rez .4 .7
AUTHOR
Hans Mikelson October 1998 New in Csound version 3.49
Page 50-13
50.9
svfilter
svfilter asig, kcf, kq[, iscl]
alow, ahigh, aband
DESCRIPTION
Implementation of a resonant second order filter, with simultaneous lowpass, highpass and bandpass outputs.
iscl coded scaling factor, similar to that in reson. A non-zero value signifies a peak response factor of 1, i.e. all frequencies other than kcf are attenuated in accordance with the (normalized) response curve. A zero value signifies no scaling of the signal, leaving that to some later adjustment (see balance). The default value is 0.
PERFORMANCE
svfilter is a second order state-variable filter, with k-rate controls for cutoff frequency and Q. As Q is increased, a resonant peak forms around the cutoff frequency. svfilter has simultaneous lowpass, highpass, and bandpass filter outputs; by mixing the outputs together, a variety of frequency responses can be generated. The state-variable filter, or multimode filter was a common feature in early analog synthesizers, due to the wide variety of sounds available from the interaction between cutoff, resonance, and output mix ratios. svfilter is well suited to the emulation of analog sounds, as well as other applications where resonant filters are called for. asig Input signal to be filtered. kcf Cutoff or resonant frequency of the filter, measured in Hz. kq Q of the filter, which is defined (for bandpass filters) as bandwidth/cutoff. kq should be in a range between 1 and 500. As kq is increased, the resonance of the filter increases, which corresponds to an increase in the magnitude and sharpness of the resonant peak. When using svfilter without any scaling of the signal (where iscl is either absent or 0), the volume of the resonant peak increases as Q increases. For high values of Q, it is recommended that iscl be set to a non-zero value, or that an external scaling function such as balance is used. svfilter is based upon an algorithm in Hal Chamberlins Musical Applications of Microprocessors (Hayden Books, 1985).
E X A M PL E
; Orchestra file for resonant filter sweep of a sawtooth-like waveform. ; The separate outputs of the filter are scaled by values from the score, ; and are mixed together. sr = 44100 = 2205 kr ksmps = 20 nchnls = 1 instr 1 idur ifreq iamp ilowamp ihighamp = = = = = p3 p4 p5 p6 p7
; determines amount of lowpass output in signal ; determines amount of highpass output in signal Signal Modifiers: Standard Filters Page 50-14
ibandamp = p8 iq = p9 iharms asig kfreq = gbuzz linseg
; determines amount of bandpass output in signal ; value of q (sr*.4) / ifreq 1, ifreq, iharms, 1, .9, 1 ; Sawtooth-like ; waveform 1, idur * 0.5, 4000, idur * 0.5, 1 ; Envelope to control ; filter cutoff svfilter asig, kfreq, iq
alow, ahigh, aband aout1 aout2 aout3 asum kenv = = = = linseg out endin
alow * ilowamp ahigh * ihighamp aband * ibandamp aout1 + aout2 + aout3 0, .1, iamp, idur -.2, iamp, .1, 0 asum * kenv
; Simple amplitude ; envelope
; Score file for above f1 0 8192 9 1 1 .25 i1 i1 i1 i1 i1 i1 i1 e 0 5 10 15 20 25 30 5 5 5 5 5 5 5 100 200 100 200 100 200 200 1000 1000 1000 1000 1000 1000 2000 1 0 0 5 1 0 0 30 0 1 0 5 0 1 0 30 0 0 1 5 0 0 1 30 .4 .6 0 ; ; ; ; ; ; ; ; lowpass sweep lowpass sweep, octave higher, higher q highpass sweep highpass sweep, octave higher, higher q bandpass sweep bandpass sweep, octave higher, higher q notch sweep - notch formed by combining highpass and lowpass outputs
AUTHOR
Page 50-15
50.10 h i l b e r t
ar1, ar2 hilbert asig
DESCRIPTION
An IIR implementation of a Hilbert transformer.
PERFORMANCE
asig input signal ar1 cosine output of asig ar2 sine output of asig hilbert is an IIR filter based implementation of a broad-band 90 degree phase difference network. The input to hilbert is an audio signal, with a frequency range from 15 Hz to 15 kHz. The outputs of hilbert have an identical frequency response to the input (i.e. they sound the same), but the two outputs have a constant phase difference of 90 degrees, plus or minus some small amount of error, throughout the entire frequency range. The outputs are in quadrature. hilbert is useful in the implementation of many digital signal processing techniques that require a signal in phase quadrature. ar1 corresponds to the cosine output of hilbert, while ar2 corresponds to the sine output. The two outputs have a constant phase difference throughout the audio range that corresponds to the phase relationship between cosine and sine waves. Internally, hilbert is based on two parallel 6th-order allpass filters. Each allpass filter implements a phase lag that increases with frequency; the difference between the phase lags of the parallel allpass filters at any given point is approximately 90 degrees. Unlike an FIR-based Hilbert transformer, the output of hilbert does not have a linear phase response. However, the IIR structure used in hilbert is far more efficient to compute, and the nonlinear phase response can be used in the creation of interesting audio effects, as in the second example below.
E X A M PL E S
The first example implements frequency shifting, or single sideband amplitude modulation. Frequency shifting is similar to ring modulation, except the upper and lower sidebands are separated into individual outputs. By using only one of the outputs, the input signal can be detuned, where the harmonic components of the signal are shifted out of harmonic alignment with each other, e.g. a signal with harmonics at 100, 200, 300, 400 and 500 Hz, shifted up by 50 Hz, will have harmonics at 150, 250, 350, 450, and 550 Hz.
sr kr ksmps nchnls = = = = 44100 4410 10 2 instr 1 idur ibegshift iendshift = = = p3 p4 p5 ; ; ; ; initial amount of frequency shiftcan be positive or negative final amount of frequency shiftcan be positive or negative
Page 50-16
kfreq
linseg
ibegshift, idur, iendshift ; A simple envelope ; for determining ; the amount of ; frequency shift. supertest.wav ; Use the sound of your choice. ain ; Phase quadrature output derived from ; input signal. ; Quadrature oscillator.
ain areal, aimag
soundin hilbert
asin acos amod1 amod2
oscili oscili = =
1, kfreq, 1 1, kfreq, 1, .25 areal * acos aimag * asin
; Trigonometric identity; see references for further ; details.
aupshift adownshift
= =
(amod1 + amod2) * 0.7 (amod1 - amod2) * 0.7
; ; ; ; ; ; ; ; ; ; ;
Both sum and difference frequencies can be output at once. aupshift corresponds to the sum frequencies, while adownshift corresponds to the difference frequencies. Notice that the adding of the two together is identical to the output of ring modulation.
aupshift, aupshift outs endin ; a simple score f1 0 16384 10 1 i1 0 29 0 200 i1 30 29 0 -200 e ; ; ; ; ; sine table for quadrature oscillator starting with no shift, ending with all frequencies shifted up by 200 Hz. starting with no shift, ending with all frequencies shifted up by 200 Hz.
The second example is a variation of the first, but with the output being fed back into the input. With very small shift amounts (i.e. between 0 and +-6 Hz), the result is a sound that has been described as a barberpole phaser or Shepard tone phase shifter. Several notches appear in the spectrum, and are constantly swept in the direction opposite that of the shift, producing a filtering effect that is reminiscent of Rissets endless glissando.
sr kr ksmps nchnls = = = = 44100 44100 1 2 instr 2 afeedback idur ibegshift iendshift ifeed init = = = = 0 p3 p4 p5 p6 ; initialization of feedback ; ; ; ; ; ; ; ; ; ; initial amount of frequency shift can be positive or negative final amount of frequency shift - can be positive or negative amount of feedback - the higher the number, the more pronounced the effect. Experiment to see at what point oscillation occurs (often a factor of 1.4 is the maximum feedback before oscillation). ; kr MUST be set to sr for barberpole effect
kfreq
linseg
ibegshift, idur, iendshift Signal Modifiers: Standard Filters Page 50-17
ain areal, aimag asin acos amod1 amod2 aupshift adownshift afeedback
soundin hilbert oscili oscili = = = = = outs endin
supertest.wav ain + afeedback 1, kfreq, 1 1, kfreq, 1, .25 areal * acos aimag * asin (amod1 + amod2) * 0.7 (amod1 - amod2) * 0.7 (amod1 - amod2) * .5 * ifeed ; feedback taken from ; downshift output aupshift, aupshift
; a simple score f1 0 16384 10 1 i2 0 29 -.3 -.3 1.4 i2 30 30 i2 60 29 .1 5 .1 1.4 -5 1.4
; ; ; ; ; ; ; ; ;
sine table for quadrature oscillator upwards sweep, at a rate of .3 times a second, lots of feedback downwards sweep, .3 times a second, lots of feedback sweep goes from .3 time a second, descending in pitch, to .3 times a second ascending in pitch, with a large amount of feedback.
TECHNICAL HISTORY
The use of phase-difference networks in frequency shifters was pioneered by Harald Bode [1]. Bode and Bob Moog provide an excellent description of the implementation and use of a frequency shifter in the analog realm [2]. This would be an excellent first source for those that wish to explore the possibilities of single sideband modulation. Bernie Hutchins provides more applications of the frequency shifter, as well as a detailed technical analysis [3]. A recent paper by Scott Wardle [4] describes a digital implementation of a frequency shifter, as well as some unique applications.
REFERENCES
1. H. Bode, Solid State Audio Frequency Spectrum Shifter. AES Preprint No. 395 (1965). 2. H. Bode and R.A. Moog, A High-Accuracy Frequency Shfiter for Professional Audio Applications. Journal of the Audio Engineering Society, July/August 1972, vol. 20, no. 6, p. 453. 3. B. Hutchins. Musical Engineers Handbook (Ithaca, NY: Electronotes, 1975), ch. 6a. 4. S. Wardle, A Hilbert-Transformer Frequency Shifter for Audio. Available online at http://www.iua.upf.es/dafx98/papers/.
AUTHOR
Page 50-18
50.11 b u t t e r h p , b u t t e r l p , b u t t e r b p ,
butterbr
ar ar ar ar butterhp butterlp butterbp butterbr asig, asig, asig, asig, kfreq [,iskip] kfreq [,iskip] kfreq, kband [,iskip] kfreq, kband [,iskip]
DESCRIPTION
Implementations of second-order high-pass, low-pass, band-pass and band-reject Butterworth filters. Note: these opcodes can also be written butlp, buthp, butbp, butbr.
PERFORMANCE
These filters are Butterworth second-order IIR filters. They are slightly slower than the original filters in Csound, but they offer an almost flat passband and very good precision and stopband attenuation. asig Input signal to be filtered. kfreq Cutoff or center frequency for each of the filters. In the case of butterbp and butterbr, the center kfreq is the intervalic, not the mathematical center. kband Bandwidth of the bandpass and bandreject filters. iskip Skip initialization if present and non zero
E X A M PL E
asig alpf ahpf abpf abrf rand butterlp butterhp butterbp butterbr 10000 asig, asig, asig, asig, 1000 500 1000, 2000 200, 150 ; ; ; ; ; White noise signal cutting frequencies above1K passing frequencies above 500Hz passing 2 octaves: 500 to 2000 Hz cutting 2 octaves: 50 to 200 Hz
AUTHOR
Page 50-19
50.12 f i l t e r 2 , z f i l t e r 2
ar kr ar filter2 filter2 zfilter2 asig, iM,iN,ib0,ib1,..., ibM,ia1,ia2,...,iaN ksig, iM,iN,ib0,ib1,...,ibM,ia1,ia2,...,iaN asig, kdamp,kfreq,iM,iN,ib0,ib1,...,ibM,ia1,ia2,...,iaN
DESCRIPTION
General purpose custom filter with time-varying pole control. The filter coefficients implement the following difference equation: (1)*y(n)=b0*x[n]+b1*x[n-1]+...+bM*x[n-M]-a1*y[n-1]-...-aN*y[n-N] the system function for which is represented by:
H(Z) =
B(Z) ---- = A(Z)
b0 + b1*Z-1 + ... + bM*Z-M --------------------------------+ ... + aN*Z-N 1 + a1*Z-1
At initialization the number of zeros and poles of the filter are specified along with the corresponding zero and pole coefficients. The coefficients must be obtained by an external filter-design application such as Matlab and specified directly or loaded into a table via GEN01. With zfilter2, the roots of the characteristic polynomials are solved at initialization so that the pole-control operations can be implemented efficiently.
PERFORMANCE
The filter2 opcodes perform filtering using a transposed form-II digital filter lattice with no time-varying control. zfilter2 uses the additional operations of radial pole-shearing and angular pole-warping in the Z plane. Pole shearing increases the magnitude of poles along radial lines in the Z-plane. This has the affect of altering filter ring times. The k-rate variable kdamp is the damping parameter. Positive values (0.01 to 0.99) increase the ring-time of the filter (hi-Q), negative values (-0.01 to -0.99) decrease the ring-time of the filter, (lo-Q). Pole warping changes the frequency of poles by moving them along angular paths in the Z plane. This operation leaves the shape of the magnitude response unchanged but alters the frequencies by a constant factor (preserving 0 and p). The k-rate variable kfreq determines the frequency warp factor. Positive values (0.01 to 0.99) increase frequencies toward p and negative values (-0.01 to -0.99) decrease frequencies toward 0. Since filter2 implements generalized recursive filters, it can be used to specify a large range of general DSP algorithms. For example, a digital waveguide can be implemented for musical instrument modeling using a pair of delayr and delayw opcodes in conjunction with the filter2 opcode.
Page 50-20
E X A M PL E
A first-order linear-phase lowpass linear-phase FIR filter operating on a k-rate signal:
k1 filter2 ksig, 2, 0, 0.5, 0.5 ;; k-rate FIR filter
A controllable second-order IIR filter operating on an a-rate signal:

; IIR filter a1 zfilter2 asig, kdamp, kfreq, 1, 2, 1, ia1, ia2 ; controllable a-rate
DEPRECATED NAMES
The k-rate version of filter2 was originally called kfilter2. As of Csound version 3.493, this name is deprecated. filter2 should be used instead of kfilter2. The opcode determines its rate from the output argument.
AUTHOR
Michael A. Casey MIT Cambridge, Mass. 1997
Page 50-21
50.13 l p f 1 8
ar lpf18 asig, kfco, kres, kdist
DESCRIPTION
Implementation of a 3 pole sweepable resonant lowpass filter.
PERFORMANCE
kfco the filter cut-off frequency in Hz. Should be in the range 0 to sr/2. kres the amount of resonance. Self-oscillation occurs when kres is approximately 1. Should usually be in the range 0 to 1, however, values slightly greater than 1 are possible for more sustained oscillation and an "overdrive" effect. kdist amount of distortion. kdist = 0 gives a clean output. kdist > 0 adds tanh() distortion controlled by the filter parameters, in such a way that both low cutoff and high resonance increase the distortion amount. Some experimentation is encouraged. lpf18 is a digital emulation of a 3 pole (18 dB/oct.) lowpass filter capable of self-oscillation with a built-in distortion unit. It is really a 3-pole version of moogvcf, retuned, recalibrated and with some performance improvements. The tuning and feedback tables use no more than 6 adds and 6 multiplies per control rate. The distortion unit, itself, is based on a modified tanh() function driven by the filter controls. Note: This filter requires that the input signal be normalized to one.
AUTHOR
Josep M Comajuncosas Spain December, 2000 New in Csound version 4.10
Page 50-22
50.14 t b v c f
ar tbvcf asig, xfco, xres, kdist, kasym
DESCRIPTION
This opcode attempts to model some of the filter characteristics of a Roland TB303 voltage-controlled filter. Euler's method is used to approximate the system, rather than traditional filter methods. Cutoff frequency, Q, and distortion are all coupled. Empirical methods were used to try to unentwine, but frequency is only approximate as a result. Future fixes for some problems with this opcode may break existing orchestras relying on this version of tbvcf.
PERFORMANCE
asig input signal. Should be normalized to 1. xfco filter cutoff frequency. Optimum range is 10,000 to 1500. Values below 1000 may cause problems. xres resonance or Q. Typically in the range 0 to 2. kdist amount of distortion. Typical value is 2. Changing kdist significantly from 2 may cause odd interaction with xfco and xres. kasym asymmetry of resonance. Typically in the range 0 to 1.
E X A M PL E
;--------------------------------------------------------; TBVCF Test ; Coded by Hans Mikelson December, 2000 ;--------------------------------------------------------= 44100 ; Sample rate sr = 4410 ; Kontrol rate kr ksmps = 10 ; Samples/Kontrol period ; Normal stereo nchnls = 2 50, 50 zakinit instr 10 idur iamp ifqc ipanl ipanr iq idist iasym kdclck kfco ax ay ay = = = = = = = = linseg expseg vco tbvcf buthp outs endin p3 p4 cpspch(p5) sqrt(p6) sqrt sqrt(1-p6) sqrt p7 p8 p9 ; ; ; ; ; Duration Amplitude Pitch to frequency Pan left Pan right
0, .002, 1, idur-.004, 1, .002, 0 ; Declick envelope 10000, idur, 1000 1, ifqc, 2, 1 ax, kfco, iq, idist, iasym ay/1, 100 ; Frequency envelope ; Square wave ; TB-VCF ; Hi-pass
ay*iamp*ipanl*kdclck, ay*iamp*ipanr*kdclck
Page 50-23
f1 0 65536 ; TeeBee ; Sta i10 0 i10 0.3 i10 0.6 i10 0.9 i10 1.2 i10 1.5 i10 1.8 i10 2.1 i10 2.4 i10 2.7 i10 3.0 i10 3.3 i10 3.6 i10 3.9 i10 4.2 i10 4.5 i10 4.8 i10 5.1 i10 5.4 i10 5.7 i10 6.0 i10 6.3 i10 6.6 i10 6.9 i10 7.2 i10 7.5 i10 7.8 i10 8.1 i10 8.4 i10 8.7
10 1 Test Dur 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2 0.2
Amp 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767 32767
Pitch 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00 7.00
Pan .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5 .5
Q 0.0 0.8 1.6 2.4 3.2 4.0 0.0 0.8 1.6 2.4 3.2 4.0 0.0 0.8 1.6 2.4 3.2 4.0 0.0 0.8 1.6 2.4 3.2 4.0 0.0 0.8 1.6 2.4 3.2 4.0
Dist1 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0 2.0
Asym 0.0 0.0 0.0 0.0 0.0 0.0 0.25 0.25 0.25 0.25 0.25 0.25 0.5 0.5 0.5 0.5 0.5 0.5 0.75 0.75 0.75 0.75 0.75 0.75 1.0 1.0 1.0 1.0 1.0 1.0
AUTHOR
Hans Mikelson December, 2000 January, 2001 New in Csound 4.10
Page 50-24
51
SIGNAL MODIFIERS: SPECIALIZED FILTERS

nlfilt
nlfilt ain, ka, kb, kd, kL, kC
51.1
ar
DESCRIPTION
Implements the filter Y{n} =a Y{n-1} + b Y{n-2} + d Y^2{n-L} + X{n} C described in Dobson and Fitch (ICMC96)
E X A M PL E
Non-linear effect: a d C L = = = = b = 0 0.8, 0.9, 0.7 0.4, 0.5, 0.6 20
This affects the lower register most but there are audible effects over the whole range. We suggest that it may be useful for coloring drums, and for adding arbitrary highlights to notes Low Pass with non-linear: a b d C L = = = = = 0.4 0.2 0.7 0.11 20, ... 200
There are instability problems with this variant but the effect is more pronounced of the lower register, but is otherwise much like the pure comb. Short values of L can add attack to a sound. High Pass with non-linear: The range of parameters are a b d C L = = = = = 0.35 -0.3 0.95 0,2, ... 0.4 200
Signal Modifiers: Specialized Filters
Page 51-1
High Pass with non-linear: The range of parameters are a b d C L = = = = = 0.7 -0.2, ... 0.5 0.9 0.12, ... 0.24 500, 10
The high pass version is less likely to oscillate. It adds scintillation to medium-high registers. With a large delay L it is a little like a reverberation, while with small values there appear to be formant-like regions. There are arbitrary color changes and resonances as the pitch changes. Works well with individual notes. Warning: The useful ranges of parameters are not yet mapped.
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK 1997
Page 51-2
51.2
ar
pareq
pareq asig, kc, iv, iq, imode
DESCRIPTION
Implementation of Zoelzers parametric equalizer filters, with some modifications by the author. The formula for the low shelf filter is: omega = 2*pi*f/sr K = tan(omega/2) b0 = 1 + sqrt(2*V)*K + V*K^2 b1 = 2*(V*K^2 - 1) b2 = 1 - sqrt(2*V)*K + V*K^2 a0 = 1 + K/Q + K^2 a1 = 2*(K^2 - 1) a2 = 1 - K/Q + K^2 The formula for the high shelf filter is: omega = 2*pi*f/sr K = tan((pi-omega)/2) b0 = 1 + sqrt(2*V)*K + V*K^2 b1 = -2*(V*K^2 - 1) b1 = 1 - sqrt(2*V)*K + V*K^2 a0 = 1 + K/Q + K^2 a1 = -2*(K^2 - 1) a2 = 1 - K/Q + K^2 The formula for the peaking filter is: omega = 2*pi*f/sr K = tan(omega/2) b0 = 1 + V*K/2 + K^2 b1 = 2*(K^2 - 1) b2 = 1 - V*K/2 + K^2 a0 = 1 + K/Q + K^2 a1 = 2*(K^2 - 1) a2 = 1 - K/Q + K^2
iv amount of boost or cut. Positive values give boost, negative values give cut. iq Q of the filter (sqrt(.5) is no resonance)
Page 51-3
imode operating mode 0 = Peaking 1 = Low Shelving 2 = High Shelving
PERFORMANCE
kc center frequency in peaking mode, corner frequency in shelving mode. asig the incoming signal
E X A M PL E
ifc iq iv imode kfc asig aout endin instr 15 = p4 ; Center / Shelf = p5 ; Quality factor sqrt(.5) is no resonance = ampdb(p6) ; Volume Boost/Cut = p7 ; Mode 0=Peaking EQ, 1=Low Shelf, 2=High Shelf linseg ifc*2, p3, ifc/2 ; Random number source for testing rand 5000 pareq asig, kfc, iv, iq, imode ; Parametric equalization ; Output the results outs aout, aout
; SCORE: ; Sta i15 0 i15 + i15 . i15 . e
Dur 1 . . .
Fcenter 10000 5000 1000 5000
Q .2 .2 .707 .1
Boost/Cut(dB) 12 12 -12 -12
Mode 1 1 2 0
AUTHOR
Hans Mikelson December, 1998 (New in Csound version 3.50)
Page 51-4
51.3
ar
dcblock
dcblock asig[, ig]
DESCRIPTION
Implements the DC blocking filter Y[i] = X[i] X[i-1] + (igain * Y[i=1]) Based on work by Perry Cook.
igain the gain of the filter, which defaults to 0.99
PERFORMANCE
ain audio signal input
AUTHOR
Page 51-5
Page 51-6
52
SIGNAL MODIFIERS: ENVELOPE MODIFIERS

linen, linenr, envlpx, envlpxr
linen linen linenr linenr envlpx envlpx envlpxr envlpxr kamp, xamp, kamp, xamp, kamp, xamp, kamp, xamp, irise, irise, irise, irise, irise, irise, irise, irise, idur, idur, idec, idec, idur, idur, idec, idec, idec idec iatdec iatdec idec, ifn, iatss, iatdec[,ixmod] idec, ifn, iatss, iatdec[,ixmod] ifn, iatss, iatdec[, ixmod[, irind]] ifn, iatss, iatdec[, ixmod[, irind]]
52.1
kr ar kr ar kr ar kr ar
DESCRIPTION
linen apply a straight line rise and decay pattern to an input amp signal. envlpx apply an envelope consisting of 3 segments: stored function rise shape modified exponential pseudo steady state exponential decay.
linenr, envlpxr as above, except that the final segment is entered only on sensing a MIDI note release, and the note is then extended by the decay time
irise rise time in seconds. A zero or negative value signifies no rise modification. idur overall duration in seconds. A zero or negative value will cause initialization to be skipped. idec decay time in seconds. Zero means no decay. An idec > idur will cause a truncated decay. irind (optional) independence flag. If left zero, the release time (idec) will influence the extended life of the current note following a note-off. If non-zero, the idec time is quite independent of the note extension (see below). The default value is 0. ifn function table number of stored rise shape with extended guard point. iatss attenuation factor, by which the last value of the envlpx rise is modified during the notes pseudo steady state. A factor l causes an exponential growth, and < l an exponential decay. A 1 will maintain a true steady state at the last rise value. Note that this attenuation is not by fixed rate (as in a piano), but is sensitive to a notes duration. However, if iatss is negative (or if steady state < 4 k-periods) a fixed attenuation rate of abs(iatss) per second will be used. 0 is illegal. iatdec attenuation factor by which the closing steady state value is reduced exponentially over the decay period. This value must be positive and is normally of the order of .01. A large or excessively small value is apt to produce a cutoff which is audible. A zero or negative value is illegal.
Signal Modifiers: Envelope Modifiers
Page 52-1
ixmod (optional, between +- .9 or so) exponential curve modifier, influencing the steepness of the exponential trajectory during the steady state. Values less than zero will cause an accelerated growth or decay towards the target (e.g. subito piano). Values greater than zero will cause a retarded growth or decay. The default value is zero (unmodified exponential).
PERFORMANCE
Rise modifications are applied for the first irise seconds, and decay from time idur idec. If these periods are separated in time there will be a steady state during which amp will be unmodified (linen) or modified by the first exponential pattern (envlpx). If linen rise and decay periods overlap then both modifications will be in effect for that time; in envlpx that will cause a truncated decay. If the overall duration idur is exceeded in performance, the final decay will continue on in the same direction, going negative for linen but tending asymptotically to zero in envlpx. linenr is unique within Csound in containing a note-off sensor and release time extender. When it senses either a score event termination or a MIDI noteoff, it will immediately extend the performance time of the current instrument by idec seconds, then execute an exponential decay towards the factor iatdec. For two or more units in an instrument, extension is by the greatest idec. linenr, envlpxr are examples of the special Csound r units that contain a note-off sensor and release time extender. Unless made independent by irind, when each senses a score event termination or a MIDI noteoff, it will immediately extend the performance time of the current instrument by idec seconds, then begin a decay (as described above) from wherever it was at the time. These r units can also be modified by MIDI noteoff velocities (see veloffs). If the irind flag is on (non-zero), the overall performance time is unaffected by note-off and veloff data.
M U L T I PL E R
UNITS
When two or more r units occur in the same instrument it is usual to have only one of them influence the overall note duration. This is normally the master amplitude unit. Other units controlling, say, filter motion can still be sensitive to note-off commands while not affecting the duration by making them independent (irind non-zero). Depending on their own idec (release time) values, independent r units may or may not reach their final destinations before the instrument terminates. If they do, they will simply hold their target values until termination. If two or more r units are simultaneously master, note extension is by the greatest idec.
Page 52-2
53
SIGNAL MODIFIERS: AMPLITUDE MODIFIERS

rms, gain, balance
rms gain balance asig[, ihp, iskip] asig, krms[, ihp, iskip] asig, acomp[, ihp, iskip]
53.1
kr ar ar
DESCRIPTION
The rms power of asig can be interrogated, set, or adjusted to match that of a comparator signal.
ihp (optional) half-power point (in Hz) of a special internal low-pass filter. The default value is 10. iskip (optional) initial disposition of internal data space ( see reson). The default value is 0.
PERFORMANCE
rms output values kr will trace the rms value of the audio input asig. This unit is not a signal modifier, but functions rather as a signal power-gauge. gain provides an amplitude modification of asig so that the output ar has rms power equal to krms. rms and gain used together (and given matching ihp values) will provide the same effect as balance. balance outputs a version of asig, amplitude-modified so that its rms power is equal to that of a comparator signal acomp. Thus a signal that has suffered loss of power (e.g., in passing through a filter bank) can be restored by matching it with, for instance, its own source. It should be noted that gain and balance provide amplitude modification only output signals are not altered in any other respect.
E X A M PL E
asrc a1 a2 afin buzz reson reson balance 10000,440, sr/440, 1 asrc, 1000,100 a1,3000,500 a2, asrc ; ; ; ; band-limited pulse train sent through 2 filters then balanced with source
Signal Modifiers: Amplitude Modifiers
Page 53-1
53.2
ar
dam
dam asig, kthreshold, icomp1, icomp2, irtime, iftime
DESCRIPTION
This opcode dynamically modifies a gain value applied to the input sound ain by comparing its power level to a given threshold level. The signal will be compressed/expanded with different factors regarding that it is over or under the threshold.
icomp1 compression ratio for upper zone. icomp2 compression ratio for lower zone. irtime gain rise time in seconds. Time over which the gain factor is allowed to raise of one unit. iftime gain fall time in seconds. Time over which the gain factor is allowed to decrease of one unit.
PERFORMANCE
asig input signal kthreshold level of input signal which acts as the threshold. Can be changed at k-time (e.g. for ducking) Note on the compression factors: A compression ratio of one leaves the sound unchanged. Setting the ratio to a value smaller than one will compress the signal (reduce its volume) while setting the ratio to a value greater than one will expand the signal (augment its volume).
AUTHOR
Marc Resibois Belgium 1997
Page 53-2
53.3
ar
clip
clip asig, imeth, ilimit[, iarg]
DESCRIPTION
Clips an a-rate signal to a predefined limit, in a soft manner, using one of three methods.
imeth selects the clipping method. The default is 0. The methods are: 0 = Bram de Jong method (default) 1 = sine clipping 2 = tanh clipping
ilimit limiting value iarg (optional) when imeth = 0, indicates the point at which clipping starts, in the range 0 1. Not used when imeth = 1 or imeth = 2. Default is 0.5.
PERFORMANCE
asig a-rate input signal The Bram de Jong method (imeth = 0) applies the algorithm: |x| > a: |x| > 1: f(x) = sin(x) * (a+(x-a)/(1+((x-a)/(1-a))2 f(x) = sin(x) * (a+1)/2
This method requires that asig be normalized to 1. The second method (imeth = 1) is the sine clip: |x| < limit: f(x) = limit * sin(*x/(2*limit)) f(x) = limit * sin(x)
The third method (imeth = 0) is the tanh clip: |x| < limit: f(x) = limit * tanh(x/limit)/tanh(1) f(x) = limit * sin(x)
Note: Method 1 appears to be non-functional at release of Csound version 4.07.
E X A M PL E
a1 a2 asig soundin oscil clip out 25000, 1 a1+a2, 0, 30000, .75 asig
AUTHOR
John ffitch University of Bath, Codemist Ltd. Bath, UK August, 2000 New in Csound version 4.07
Page 53-3
Page 53-4
54
54.1
ir kr ar ir kr ar ir kr ar
SIGNAL MODIFIERS: SIGNAL LIMITERS

limit, mirror, wrap
limit limit limit wrap wrap wrap mirror mirror mirror isig, ilow, ihigh ksig, klow, khigh asig, klow, khigh isig, ilow, ihigh ksig, klow, khigh asig, klow, khigh isig, ilow, ihigh ksig, klow, khigh asig, klow, khigh
DESCRIPTION
Wraps the signal in various ways.
isig input signal ilow low threshold ihigh high threshold
PERFORMANCE
xsig input signal xlow low threshold xhigh high threshold limit sets lower and upper limits on the xsig value they process. If xhigh is lower than xlow, then the output will be the average of the two it will not be affected by xsig. mirror reflects the signal that exceeds the low and high thresholds. wrap wraps-around the signal that exceeds the low and high thresholds. These opcodes are useful in several situations, such as table indexing or for clipping and modeling i-rate, k-rate or a-rate signals. wrap is also useful for wrap-around of table data when the maximum index is not a power of two (see table and tablei). Another use of wrap is in cyclical event repeating, with arbitrary cycle length.
Signal Modifiers: Signal Limiters
Page 54-1
AUTHORS
Gabriel Maldonado (wrap, mirror) Italy New in Csound version 3.49 Robin Whittle (limit) Australia New in Csound version 3.46
Page 54-2
55
55.1
ar ar ar
SIGNAL MODIFIERS: DELAY

delayr, delayw, delay, delay1
delayr delayw delay delay1 idlt[, iskip] asig asig, idlt[, iskip] asig[, iskip]
DESCRIPTION
A signal can be read from or written into a delay path, or it can be automatically delayed by some time interval.
idlt requested delay time in seconds. This can be as large as available memory will permit. The space required for n seconds of delay is 4n * sr bytes. It is allocated at the time the instrument is first initialized, and returned to the pool at the end of a score section. iskip (optional) initial disposition of delay-loop data space ( see reson). The default value is 0.
PERFORMANCE
delayr reads from an automatically established digital delay line, in which the signal retrieved has been resident for idlt seconds. This unit must be paired with and precede an accompanying delayw unit. Any other Csound statements can intervene. delayw writes asig into the delay area established by the preceding delayr unit. Viewed as a pair, these two units permit the formation of modified feedback loops, etc. However, there is a lower bound on the value of idlt, which must be at least 1 control period (or 1/kr). delayr/delayw pairs may be interleaved. Beginning another delayr/delayw pair before terminating a previous pair is no longer excluded. For the interleaved pairs, the first delayr unit is associated with the first delayw unit, the second delayr unit with the second delayw unit, and so on. In this way, it is possible to implement cross-coupled feedback that is completed within the same control-rate cycle. See Example 2. (This feature added in Csound version 3.57 by Jens Groh and John ffitch). delay is a composite of the above two units, both reading from and writing into its own storage area. It can thus accomplish signal time-shift, although modified feedback is not possible. There is no minimum delay period. delay1 is a special form of delay that serves to delay the audio signal asig by just one sample. It is thus functionally equivalent to delay asig, 1/sr but is more efficient in both time and space. This unit is particularly useful in the fabrication of generalized nonrecursive filters.
Signal Modifiers: Delay
Page 55-1
E X A M PL E S
Example 1:
tigoto a2 delay contin: contin a1, .05, 0 ; except on a tie, ; begin 50 msec clean delay of sig
Example 2:
ainput1 ainput2 = = ..... .....
;Read delayed signal, first delayr instance: adly1 delayr 0.11 ;Read delayed signal, second delayr instance: adly2 delayr 0.07 ;Do some cross-coupled manipulation: afdbk1 = 0.7 * adly1 + 0.7 * adly2 + ainput1 afdbk2 = -0.7 * adly1 + 0.7 * adly2 + ainput2 ;Feed back signal associated with first delayr instance: afdbk1 delayw ;Feed back signal associated with second delayr instance: afdbk2 delayw adly1, adly2 outs
Page 55-2
55.2
ar ar ar ar
deltap, deltapi, deltapn, deltap3

deltap deltapi deltapn deltap3 kdlt xdlt xnumsamps xdlt
DESCRIPTION
Tap a delay line at variable offset times.
PERFORMANCE
These units can tap into a delayr/delayw pair, extracting delayed audio from the idlt seconds of stored sound. There can be any number of deltap and/or deltapi units between a read/write pair. Each receives an audio tap with no change of original amplitude. deltap extracts sound by reading the stored samples directly; deltapi extracts sound by interpolated readout. By interpolating between adjacent stored samples deltapi represents a particular delay time with more accuracy, but it will take about twice as long to run. The arguments kdlt, xdlt specify the tapped delay time in seconds. Each can range from 1 Control Period to the full delay time of the read/write pair; however, since there is no internal check for adherence to this range, the user is wholly responsible. Each argument can be a constant, a variable, or a time-varying signal; the xdlt argument in deltapi implies that an audio-varying delay is permitted there. deltapn is identical to deltapi, except delay time is specified in number of samples, instead of seconds (Hans Mikelson). deltap3 is experimental, and uses cubic interpolation. (New in Csound version 3.50.) These units can provide multiple delay taps for arbitrary delay path and feedback networks. They can deliver either constant-time or time-varying taps, and are useful for building chorus effects, harmonizers, and Doppler shifts. Constant-time delay taps (and some slowly changing ones) do not need interpolated readout; they are well served by deltap. Medium-paced or fast varying dlts, however, will need the extra services of deltapi. delayr/delayw pairs may be interleaved. To associate a delay tap unit with a specific delayr unit, it not only has to be located between that delayr and the appropriate delayw unit, but must also precede any following delayr units. See Example 2. (This feature added in Csound version 3.57 by Jens Groh and John ffitch). N.B. k-rate delay times are not internally interpolated, but rather lay down stepped timeshifts of audio samples; this will be found quite adequate for slowly changing tap times. For medium to fast-paced changes, however, one should provide a higher resolution audiorate timeshift as input.
E X A M PL E S
Example 1:
asource atime ampfac adump amove buzz linseg = delayr deltapi delayw out 1, 440, 20, 1 1, p3/2,.01, p3/2,1 ; trace a distance in secs 1/atime/atime ; and calc an amp factor 1 ; set maximum distance atime ; move sound source past asource ; the listener amove * ampfac
Page 55-3
Example 2:
ainput1 ainput2 kdlyt1 kdlyt2 = = = = ..... ..... ..... .....
;Read delayed signal, first delayr instance: adump 4.0 delayr adly1 deltap kdlyt1 ;associated with first delayr instance ;Read delayed signal, second delayr instance: adump 4.0 delayr adly2 kdlyt2 deltap instance
; associated with second delayr
;Do some cross-coupled manipulation: afdbk1 = 0.7 * adly1 + 0.7 * adly2 + ainput1 afdbk2 = -0.7 * adly1 + 0.7 * adly2 + ainput2 ;Feed back signal, associated with first delayr instance: afdbk1 delayw ;Feed back signal, associated with second delayr instance: afdbk2 delayw adly1, adly2 outs
Page 55-4
55.3
ar
multitap
multitap asig, itime1, igain1, itime2, igain2 . . .
DESCRIPTION
Multitap delay line implementation.
The arguments itime and igain set the position and gain of each tap. The delay line is fed by asig.
E X A M PL E
a1 a2 oscil multitap out 1000, 100, 1 a1, 1.2, .5, 1.4, .2 a2
This results in two delays, one with length of 1.2 and gain of .5, and one with length of 1.4 and gain of .2.
AUTHOR
Page 55-5
55.4
ar ar
vdelay, vdelay3
vdelay vdelay3 asig, adel, imaxdel asig, adel, imaxdel [, iskip] [, iskip]
DESCRIPTION
This is an interpolating variable time delay, it is not very different from the existing implementation (deltapi), it is only easier to use. vdelay3 is experimental, and is the same as vdelay, except that it uses cubic interpolation. (New in Version 3.50.)
imaxdel Maximum value of delay in milliseconds. If adel gains a value greater than imaxdel it is folded around imaxdel. This should not happen. iskip Skip initialization if present and non-zero
PERFORMANCE
With this unit generator it is possible to do Doppler effects or chorusing and flanging. asig Input signal. adel Current value of delay in milliseconds. Note that linear functions have no pitch change effects. Fast changing values of adel will cause discontinuities in the waveform resulting noise.
E X A M PL E
f1 0 ims a1 a2 a2 a3 8192 10 = oscil oscil = vdelay out 1 100 10000, 1737, 1 ims/2, 1/p3, 1 a2 + ims/2 a1, a2, ims a3 ; ; ; ; ; Maximum delay time in msec Make a signal Make an LFO Offset the LFO so that it is positive Use the LFO to control delay time
Two important points here. First, the delay time must be always positive. And second, even though the delay time can be controlled in k-rate, it is not advised to do so, since sudden time changes will create clicks.
AUTHOR
Page 55-6
56
56.1
ar ar ar
SIGNAL MODIFIERS: REVERBERATION

comb, alpass, reverb
comb alpass reverb asig, krvt, ilpt[, iskip][, insmps] asig, krvt, ilpt[, iskip][, insmps] asig, krvt[, iskip]
DESCRIPTION
An input signal is reverberated for krvt seconds with colored (comb), flat (alpass), or natural room (reverb) frequency response.
ilpt loop time in seconds, which determines the echo density of the reverberation. This in turn characterizes the color of the comb filter whose frequency response curve will contain ilpt * sr/2 peaks spaced evenly between 0 and sr/2 (the Nyquist frequency). Loop time can be as large as available memory will permit. The space required for an n second loop is 4n* sr bytes. comb and alpass delay space is allocated and returned as in delay. iskip (optional) initial disposition of delay-loop data space ( cf. reson). The default value is 0. insmps (optional) if non-zero, loop time is in samples instead of seconds. Default is zero. New in Csound version 4.10.
PERFORMANCE
These filters reiterate input with an echo density determined by loop time ilpt. The attenuation rate is independent and is determined by krvt, the reverberation time (defined as the time in seconds for a signal to decay to 1/1000, or 60dB down from its original amplitude). Output from a comb filter will appear only after ilpt seconds; alpass output will begin to appear immediately. A standard reverb unit is composed of four comb filters in parallel followed by two alpass units in series. Loop times are set for optimal natural room response. Core storage requirements for this unit are proportional only to the sampling rate, each unit requiring approximately 3K words for every 10KC. The comb, alpass, delay, tone and other Csound units provide the means for experimenting with alternate reverberator designs Since output from the standard reverb will begin to appear only after 1/20 second or so of delay, and often with less than three-fourths of the original power, it is normal to output both the source and the reverberated signal. If krvt is inadvertently set to a non-positive number, krvt will be reset automatically to 0.01. (New in Csound version 4.07.) Also, since the reverberated sound will persist long after the cessation of source events, it is normal to put reverb in a separate instrument to which sound is passed via a global variable, and to leave that instrument running throughout the performance.
Signal Modifiers: Reverberation
Page 56-1
E X A M PL E
ga1 a1 ga1 a3 ga1 init instr oscili out = endin instr reverb out = endin 0 ; init an audio receiver/mixer 1 ; instr (there may be many copies) 8000, cpspch(p5), 1 ; generate a source signal a1 ; output the direct sound ga1 + a1 ; and add to audio receiver 99 ga1, 1.5 a3 0 ; ; ; ; (highest instr number executed last) reverberate whatever is in ga1 and output the result empty the receiver for the next pass
Page 56-2
56.2
ar ar
reverb2, nreverb
reverb2 nreverb asig, ktime, khdif [,iskip] asig, ktime, khdif [,iskip] [,inumCombs, ifnCombs]\\ [, inumAlpas, ifnAlpas]
DESCRIPTION
This is a reverberator consisting of 6 parallel comb-lowpass filters being fed into a series of 5 allpass filters. nreverb replaces reverb2 (version 3.48) and so both opcodes are identical.
INITIALISATION
iskip Skip initialization if present and non zero inumCombs number of filter constants in comb filter. If omitted, the values default to the nreverb constants. New in Csound version 4.09. ifnCombs function table with inumCombs comb filter time values, followed the same number of gain values. The ftable should not be rescaled (use negative fgen number). Positive time values are in seconds. The time values are converted internally into number of samples, then set to the next greater prime number. If teh time is negative, it is interpreted directly as time in sample frames, and no processing is done (except negation). New in Csound version 4.09. inumAlpas, ifnAlpas same as inumCombs and ifnCombs, for allpass filter. New in Csound version 4.09.
PERFORMANCE
The output of nreverb (and reverb2) is zeroed on the first perfomance pass. The input signal asig is reverberated for ktime seconds. The parameter khdif controls the high frequency diffusion amount. The values of khdif should be from 0 to 1. If khdif is set to 0 the all the frequencies decay with the same speed. If khdif is 1, high frequencies decay faster than lower ones. If ktime is inadvertently set to a non-positive number, ktime will be reset automatically to 0.01. (New in Csound version 4.07.) As of Csound version 4.09, nreverb may read any number of comb and allpass filter from an ftable.
E X A M PL E S
This results in a 2.5 sec reverb with faster high frequency attenuation:
a1 a2
10000, 100, 1 oscil reverb2 a1, 2.5, .3 a1 + a2 * .2 out
Page 56-3
This illustrates the use of an ftable for filter constants:

;Orchestra: a1 a2
instr 1 soundin nreverb out endin
"neopren.wav" a1, 1.5, .75, 0, 8, 71, 4, 72 a1 + a2 * .4
;Score: ; freeverb time constants, as direct (negative) sample, with arbitrary gains f71 0 16 -2 -1116 -1188 -1277 -1356 -1422 -1491 -1557 -1617 0.8 0.79 0.78 \\ 0.77 0.76 0.75 0.74 0.73 f72 0 16 -2 -556 -441 -341 -225 0.7 0.72 0.74 0.76 i1 e 0 7
AUTHORS
Paris Smaragdis (reverb2) MIT, Cambridge 1995 Richard Karpen (nreverb) Seattle, Wash 1998
Page 56-4
56.3
ar
nestedap
nestedap asig, imode, imaxdel, idel1, igain1[, idel2, igain2\\ [, idel3, igain3]]
DESCRIPTION
Three different nested all-pass filters, useful for implementing reverbs.
imode operating mode of the filter: 1 = simple all-pass filter 2 = single nested all-pass filter 3 = double nested all-pass filter idel1, idel2, idel3 delay times of the filter stages. Delay times are in seconds and must be greater than zero. idel1 must be greater than the sum of idel2 and idel3. igain1, igain2, igain3 gain of the filter stages. imaxdel will be necessary if k-rate delays are implemented. Not currently used.
PERFORMANCE
asig input signal If imode = 1, the filter takes the form: id el
If imode = 2, the filter takes the form: id el idel2 ,
Page 56-5
If imode = 3, the filter takes the form:
idel1, igain1
E X A M PL E
insnd gasig instr 5 = p4 diskin endin instr 10 = = = = = = = = = = = = = init nestedap igain3 nestedap nestedap butterlp outs gasig = endin ;Score f1 0 8192 10 1 ; Diskin ; Sta Dur i5 0 3 ; Reverb ; St Dur i10 0 4 e Soundin 1 Del1 Gn1 97 .11 Del2 Gn2 23 .07 Del3 Gn3 43 .09 Del4 72 Gn4 .2 Del5 53 Gn5 .2 Del6 119 Gn6 .3 insnd, 1
imax idel1 igain1 idel2 igain2 idel3 igain3 idel4 igain4 idel5 igain5 idel6 igain6 afdbk aout1 idel3, aout2 aout afdbk
1 p4 p5 p6 p7 p8 p9 p10 p11 p12 p13 p14 p15 0 gasig+afdbk*.4, 3, imax, idel1, igain1, idel2,\\ aout1, 2, imax, idel4, igain4, idel5, igain5 aout2, 1, imax, idel6, igain6 aout, 1000 gasig+(aout+aout1)/2, gasig-(aout+aout1)/2 0 igain2,
AUTHOR
Hans Mikelson February 1999 New in Csound version 3.53
Page 56-6
56.4
a1, a2
babo
babo asig, ksrcx, ksrcy, ksrcz, irx, iry, irz[, idiff\\ [, ifno]]
DESCRIPTION
babo stands for ball-within-the-box. It is a physical model reverberator based on a paper by Davide Rocchesso "The Ball within the Box: a sound-processing metaphor", Computer Music Journal, Vol 19, N.4, pp.45-47, Winter 1995. The resonator geometry can be defined, along with some response characteristics, the position of the listener within the resonator, and the position of the sound source.
irx, iry, irz the coordinates of the geometry of the resonator (length of the edges in meters) idiff is the coefficient of diffusion at the walls, which regulates the amount of diffusion (0-1, where 0 = no diffusion, 1 = maximum diffusion - default: 1) ifno expert values function: a function number that holds all the additional parameters of the resonator. This is typically a GEN2--type function used in non-rescaling mode. They are as follows: decay main decay of the resonator (default: 0.99) hydecay high frequency decay of the resonator (default: 0.1) rcvx, rcvy, rcvz the coordinates of the position of the receiver (the listener) (in meters; 0,0,0 is the resonator center) rdistance the distance in meters between the two pickups (your ears, for example - default: 0.3) direct the attenuation of the direct signal (0-1, default: 0.5) early_diff the attenuation coefficient of the early reflections (0-1, default: 0.8)
PERFORMANCE
asig the input signal ksrcx, ksrcy, ksrcz the virtual coordinates of the source of sound (the input signal). These are allowed to move at k-rate and provide all the necessary variations in terms of response of the resonator.
E X A M PL E S
Orchestra File - Simple usage
; minimal babo instrument ; instr 1 ix =p5 ; x position of iy =p6 ; y position of iz =p7 ; z position of ixsize =p8 ; width of the iysize =p9 ; depth of the izsize =p10 ; height of the ainput al,ar soundin p4 babo outs ainput*0.9, ix, iy, iz, ixsize, iysize, izsize al,ar
source source source resonator resonator resonator
endin
The Public Csound Reference Manual Version 4.10 Signal Modifiers: Reverberation Page 56-7
Score File - Simple Usage

; simple babo usage: ; ;p4 : soundin number ;p5 : x position of source ;p6 : y position of source ;p7 : z position of source ;p1 : width of the resonator ;p12 : depth of the resonator ;p13 : height of the resonator ; i1 0 10 1 6 4 3 14.39 11.86 10 ; ^^^^^^^ ^^^^^^^^^^^^^^ ; ||||||| ++++++++++++++: optimal room dims according to ; ||||||| Milner and Bernard JASA 85(2), 1989 ; +++++++++: source position e
Orchestra File - Expert usage

; full blown babo instrument with movement ; instr 2 ixstart=p5 ; start x position of source (left-right) ixend =p8 ; end x position of source iystart=p6 ; start y position of source (front-back) iyend =p9 ; end y position of source izstart=p7 ; start z position of source (up-down) izend =p10 ; end z position of source ixsize =p11 ; width of the resonator iysize =p12 ; depth of the resonator izsize =p13 ; height of the resonator idiff =p14 ; diffusion coefficient iexpert=p15 ; power user values stored in this function ainput ksource_x ksource_y ksource_z soundin line line line p4 ixstart, p3, ixend iystart, p3, iyend izstart, p3, izend
al,ar babo ainput*0.9, ksource_x, ksource_y, ksource_z, ixsize, iysize, izsize, idiff, iexpert outs endin Score File - Expert Usage ; full blown instrument ;p5 : start x position of source (left-right) ;p6 : end x position of source ;p7 : start y position of source (front-back) ;p8 : end y position of source ;p9 : start z position of source (up-down) ;p10 : end z position of source ;p11 : width of the resonator ;p12 : depth of the resonator ;p13 : height of the resonator ;p14 : diffusion coefficient ;p15 : power user values stored in this function ; f1 f2 decay hidecay rx ry rz rdistance direct early_diff 0 8 -2 0.95 0.95 0 0 0 0.3 0.5 0.8 ; brighter 0 8 -2 0.95 0.5 0 0 0 0.3 0.5 0.8 ; default (to be set 8 -2 8 -2 8 -2 0.95 0.95 0.9 0.01 0.7 0.5 0 0 0 0 0 0 0 0 0 0.3 0.3 0.3 0.5 0.1 2.0 0.8 0.4 ; darker ; to hear the effect al,ar
as)
f3 0 f4 0 of diffusion f5 0 movement f6 0 ; ;
0.98 ; to hear the ; default vals
8 -2 0.99 0.1 0 0 0 0.3 0.5 0.8 ^ ----- gen. number: negative to avoid rescaling
Page 56-8
i2 i2 i2 i2 i2 i2 i2 ; i2 i2 i2 i2 i2 i2 ; ; ; ; ; ; e
10 10 1 6 + 4 2 6 + 4 2 6 + 4 2 6 + 3 2 .6 + 3 2 .6 + 4 2 12 + + + + + + 4 4 4 3 3 4
4 4 4 4 .4 .4 4
3 6 3 6 3 -6 3 -6 .3 -.6 .3 -.6 3 -12
4 3 4 3 -4 3 -4 3 -.4 .3 -.4 .3 -4 -3
14.39 14.39 14.39 14.39 1.439 1.439 24.39
11.86 11.86 11.86 11.86 1.186 1.186 21.86
10 10 10 10 1.0 1.0 20
1 1 1 1 0.0 1.0 1
6 1 2 3 4 4 5
; ; ; ; ; ; ;
defaults hear brightness 1 hear brightness 2 hear brightness 3 hear diffusion 1 hear diffusion 2 hear movement
1 6 4 3 6 4 3 1 6 4 3 -6 -4 3 1 6 4 3 -6 -4 3 1 .6 .4 .3 -.6 -.4 .3 1 .6 .4 .3 -.6 -.4 .3 1 12 4 3 -12 -4 -3 ^^^^^^^^^^^^^^^^^^^ ||||||||||||||||||| ||||||||||||||||||| |||||||||||||||||||
||||||||||||||||||| --------------------: source position start and end
14.39 11.86 10 1 1 ; hear brightness 1 14.39 11.86 10 1 2 ; hear brightness 2 14.39 11.86 10 1 3 ; hear brightness 3 1.439 1.186 1.0 0.0 4 ; hear diffusion 1 1.439 1.186 1.0 1.0 4 ; hear diffusion 2 24.39 21.86 20 1 5 ; hear movement ^^^^^^^^^^^^^^^^^ ^ ^ ||||||||||||||||| |-: expert values function ||||||||||||||||| +--: diffusion optimal room dims according to Milner and Bernard JASA 85(2), 1989
AUTHORS
Paolo Filippi Padova, Italy 1999 Nicola Bernardini Rome, Italy 2000 New in Csound version 4.09
Page 56-9
Page 56-10
57
57.1
ar ar
SIGNAL MODIFIERS: WAVEGUIDES

wguide1, wguide2
wguide1 wguide2 asig, xfreq, xcutoff, kfeedback; asig, xfreq1, xfreq2, kcutoff1, kcutoff2, kfeedback1,\ kfeedback2
DESCRIPTION
Simple waveguide blocks
PERFORMANCE
asig the input of excitation noise xfreq the frequency (i.e. the inverse of delay time) Changed to x-rate in Csound version 3.59. kcutoff1, kcutoff2 the filter cutoff frequency in Hz xcutoff the filter cutoff frequency in Hz. Changed to x-rate for wguide1 in Csound version 3.59. kfeedback the feedback factor wguide1 is the most elemental waveguide model, consisting of one delayline and one firstorder lowpass filter. wguide2 is a model of beaten plate consisting of two parallel delaylines and two first-order lowpass filters. The two feedback lines are mixed and sent to the delay again each cycle. Implementing waveguide algorithms as opcodes, instead of as orc instr, allows the user to set kr different than sr, allowing better performance particularly when using real-time.
wguide1
Signal Modifiers: Waveguides
Page 57-1
wguide2
AUTHOR
Gabriel Maldonado Italy October, 1998 (New in Csound version 3.49)
Page 57-2
57.2
ar
streson
streson asig, kfr, ifdbgain
An audio signal is modified by a string resonator with variable fundamental frequency.
ifdbgain feedback gain, between 0 and 1, of the internal delay line. A value close to 1 creates a slower decay and a more pronounced resonance. Small values may leave the input signal unaffected. Depending on the filter frequency, typical values are > .9.
PERFORMANCE
streson passes the input asig through a network composed of comb, low-pass and all-pass filters, similar to the one used in some versions of the Karplus-Strong algorithm, creating a string resonator effect. The fundamental frequency of the string is controlled by the krate variable kfr. This opcode can be used to simulate sympathetic resonances to an input signal. streson is an adaptation of the StringFlt object of the SndObj Sound Object Library developed by the author.
AUTHOR
Victor Lazzarini Music Department National University of Ireland, Maynooth Maynooth, Co. Kildare 1998 (New in Csound version 3.494)
Page 57-3
Page 57-4
58
58.1
ar
SIGNAL MODIFIERS: SPECIAL EFFECTS

harmon
harmon asig, kestfrq, kmaxvar, kgenfreq1, kgenfreq2, imode,\\ iminfrq, iprd
DESCRIPTION
Analyze an audio input and generate harmonizing voices in synchrony.
imode interpreting mode for the generating frequency inputs kgenfreq1, kgenfreq2. 0: input values are ratios with respect to the audio signal analyzed frequencies. 1: input values are the actual requested frequencies in Hz. iminfrq the lowest expected frequency (in Hz) of the audio input. This parameter determines how much of the input is saved for the running analysis, and sets a lower bound on the internal pitch tracker. iprd period of analysis (in seconds). Since the internal pitch analysis can be timeconsuming, the input is typically analyzed only each 20 to 50 milliseconds.
PERFORMANCE
This unit is a harmonizer, able to provide up to two additional voices with the same amplitude and spectrum as the input. The input analysis is assisted by two things: an input estimated frequency kestfrq (in Hz), and a fractional maximum variance kmaxvar about that estimate which serves to limit the size of the search. Once the real input frequency is determined, the most recent pulse shape is used to generate the other voices at their requested frequencies. The three frequency inputs can be derived in various ways from a score file or MIDI source. The first is the expected pitch, with a variance parameter allowing for inflections or inaccuracies; if the expected pitch is zero the harmonizer will be silent. The second and third pitches control the output frequencies; if either is zero the harmonizer will output only the non-zero request; if both are zero the harmonizer will be silent. When the requested frequency is higher than the input, the process requires additional computation due to overlapped output pulses. This is currently limited for efficiency reasons, with the result that only one voice can be higher than the input at any one time. This unit is useful for supplying a background chorus effect on demand, or for correcting the pitch of a faulty input vocal. There is essentially no delay between input and output. Output includes only the generated parts, and does not include the input.
Signal Modifiers: Special Effects
Page 58-1
E X A M PL E
asig1 kcps1 asig2 in ; get the live input cpsmidib ; and its target pitch harmon asig1, kcps1, .3, kcps1, kcps1 * 1.25, 1, 110, .04 ; add maj 3rd out asig2 ; output just the corrected and added voices
AUTHOR
Barry Vercoe MIT, Cambridge, Mass 1997
Page 58-2
58.2
ar
flanger
flanger asig, adel, kfeedback[, imaxd ]
DESCRIPTION
A user controlled flanger.
imaxd(optional) maximum delay in seconds (needed for initial memory allocation)
PERFORMANCE
asig input signal adel delay in seconds kfeedback feedback amount (in normal tasks this should not exceed 1, even if bigger values are allowed) This unit is useful for generating choruses and flangers. The delay must be varied at a-rate connecting adel to an oscillator output. Also the feedback can vary at k-rate. This opcode is implemented to allow kr different than sr (else delay could not be lower than ksmps) enhancing real-time performance. This unit is very similar to wguide1, the only difference is flanger does not have the lowpass filter.
flanger
AUTHOR
Page 58-3
58.3
ar
distort1
distort1 asig[, kpregain[, kpostgain[, kshape1[, kshape2]]]]
DESCRIPTION
Implementation of modified hyperbolic tangent distortion. distort1 can be used to generate wave shaping distortion based on a modification of the tanh function. exp(asig * (pregain + shape1)) - exp(asig*(pregain+shape2)) aout = ----------------------------------------------------------exp(asig*pregain) + exp(-asig*pregain)
PERFORMANCE
asig is the input signal. kpregain determines the amount of gain applied to the signal before waveshaping. A value of 1 gives slight distortion. kpostgain determines the amount of gain applied to the signal after waveshaping. kshape1 determines the shape of the positive part of the curve. A value of 0 gives a flat clip, small positive values give sloped shaping. kshape2 determines the shape of the negative part of the curve. All arguments except asig, were made optional in Csound version 3.52.
E X A M PL E
gadist iamp ifqc asig gadist init 0 instr 1 = p4 = cpspch(p5) iamp, ifqc, ifqc, 0, 1 pluck = gadist + asig endin instr 50 p4 init p5 init p6 init p7 init distort1 gadist, kpre, kpost, kshap1, kshap2 outs aout, aout = 0 endin Dur 3.0 2.5 2.0 1.5 Dur 3 Amp 10000 10000 10000 10000 Pitch 6.00 7.00 7.07 8.00
kpre kpost kshap1 kshap2 aout gadist ; i1 i1 i1 i1 Sta 0.0 0.5 1.0 1.5
; Sta i50 0 e
PreGain PostGain Shape1 Shape2 2 1 0 0
Page 58-4
AUTHOR
Hans Mikelson December 1998 (New in Csound version 3.50)
Page 58-5
58.4
ar ar
phaser1, phaser2
phaser1 phaser2 asig, kfreq, iord, kfeedback[, iskip] asig, kfreq, kq, iord, imode, ksep, kfeedback
DESCRIPTION
An implementation of iord number of first-order (phaser1) or second-order (phaser2) allpass filters in series.
iord the number of allpass stages in series. For phaser1, these are first-order filters, and iord can range from 1 to 4999. For phaser2, these are second-order filters, and iord can range from 1 to 2499. With higher orders, the computation time increases. iskip used to control initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0. imode used in calculation of notch frequencies.
PERFORMANCE
kfreq frequency (in Hz) of the filter(s). For phaser1, this is the frequency at which each filter in the series shifts its input by 90 degrees. For phaser2, this is the center frequency of the notch of the first allpass filter in the series. This frequency is used as the base frequency from which the frequencies of the other notches are derived. kq Q of each notch. Higher Q values result in narrow notches. A Q between 0.5 and 1 results in the strongest phasing effect, but higher Q values can be used for special effects. kfeedback amount of the output which is fed back into the input of the allpass chain. With larger amounts of feedback, more prominent notches appear in the spectrum of the output. kfeedback must be between -1 and +1. for stability. ksep scaling factor used, in conjunction with imode, to determine the frequencies of the additional notches in the output spectrum. phaser1 implements iord number of first-order allpass sections, serially connected, all sharing the same coefficient. Each allpass section can be represented by the following difference equation: y(n) = C * x(n) + x(n-1) - C * y(n-1) where x(n) is the input, x(n-1) is the previous input, y(n) is the output, y(n-1) is the previous output, and C is a coefficient which is calculated from the value of kfreq, using the bilinear z-transform. By slowly varying kfreq, and mixing the output of the allpass chain with the input, the classic phase shifter effect is created, with notches moving up and down in frequency. This works best with iord between 4 and 16. When the input to the allpass chain is mixed with the output, 1 notch is generated for every 2 allpass stages, so that with iord = 6, there will be 3 notches in the output. With higher values for iord, modulating kfreq will result in a form of nonlinear pitch modulation.
Page 58-6
phaser2 implements iord number of second-order allpass sections, connected in series. The use of second-order allpass sections allows for the precise placement of the frequency, width, and depth of notches in the frequency spectrum. iord is used to directly determine the number of notches in the spectrum; e.g. for iord = 6, there will be 6 notches in the output spectrum. There are two possible modes for determining the notch frequencies. When imode = 1, the notch frequencies are determined the following function: frequency of notch N = kbf + (ksep * kbf * N-1) For example, with imode = 1 and ksep = 1, the notches will be in harmonic relationship with the notch frequency determined by kfreq (i.e. if there are 8 notches, with the first at 100 Hz, the next notches will be at 200, 300, 400, 500, 600, 700, and 800 Hz). This is useful for generating a comb filtering effect, with the number of notches determined by iord. Different values of ksep allow for inharmonic notch frequencies and other special effects. ksep can be swept to create an expansion or contraction of the notch frequencies. A useful visual analogy for the effect of sweeping ksep would be the bellows of an accordion as it is being played the notches will be seperated, then compressed together, as ksep changes. When imode = 2, the subsequent notches are powers of the input parameter ksep times the initial notch frequency specified by kfreq. This can be used to set the notch frequencies to octaves and other musical intervals. For example, the following lines will generate 8 notches in the output spectrum, with the notches spaced at octaves of kfreq:
aphs aout phaser2 = ain, kfreq, 0.5, 8, 2, 2, 0 ain + aphs
When imode = 2, the value of ksep must be greater than 0. ksep can be swept to create a compression and expansion of notch frequencies (with more dramatic effects than when imode = 1).
E X A M PL E S
; Orchestra for demonstration of phaser1 and phaser2 = 44100 sr = 4410 kr ksmps = 10 nchnls = 1 instr 1 ; ; ; ; ; ; ; ; ; ; ; ; demonstration of phase shifting ; abilities of phaser1. Input mixed with output of phaser1 to generate notches. Shows the effects of different iorder values on the sound number of 1st-order stages in phaser1 network. Divide iorder by 2 to get the number of notches. frequency of modulation of phaser1 amount of feedback for phaser1
idur iamp iorder
= = =
p3 p4 * .05 p5
ifreq ifeed kamp iharms asig
= = linseg = gbuzz
p6 p7
0, .2, iamp, idur - .2, iamp, .2, 0 (sr*.4) / 100 1, 100, iharms, 1, .95, 2 ; "Sawtooth" waveform
; modulation oscillator for phaser1 ugen. kfreq 5500, ifreq, 1 oscili kmod = kfreq + 5600
Page 58-7
aphs
phaser1 out endin instr 2
asig, kmod, iorder, ifeed (asig + aphs) * iamp
idur iamp iorder ifreq ifeed imode isep kamp iharms asig
= = = = = = = linseg = gbuzz
p3 p4 * .04 p5 p6 p7 p8 p9
; ; ; ; ; ;
demonstration of phase shifting abilities of phaser2. Input mixed with output of phaser2 to generate notches. Demonstrates the interaction of imode and ksep.
; number of 2nd-order stages in ; phaser2 network ; ; ; ; ; not used amount of feedback for phaser2 mode for frequency scaling used with imode to determine notch frequencies
0, .2, iamp, idur - .2, iamp, .2, 0 (sr*.4) / 100 1, 100, iharms, 1, .95, 2 ; "Sawtooth" waveform
; exponentially decaying function, to control notch frequencies kline 1, idur, .005 expseg aphs phaser2 asig, kline * 2000, .5, iorder, imode, isep, ifeed (asig + aphs) * iamp out endin ; score file for above f1 0 16384 9 .5 -1 0 ; inverted half-sine, used ; for modulating phaser1 frequency f2 0 8192 9 1 1 .25 ; cosine wave for gbuzz ; phaser1 i1 0 5 7000 4 .2 .9 i1 6 5 7000 6 .2 .9 i1 12 5 7000 8 .2 .9 i1 18 5 7000 16 .2 .9 i1 24 5 7000 32 .2 .9 i1 30 5 7000 64 .2 .9 ; phaser2, imode=1 i2 37 10 7000 8 .2 .9 1 .33 i2 48 10 7000 8 .2 .9 1 2 ; phaser2, imode=2 i2 60 10 7000 8 .2 .9 2 .33 i2 72 10 7000 8 .2 .9 2 2 e
TECHNICAL HISTORY
A general description of the differences between flanging and phasing can be found in Hartmann [1]. An early implementation of first-order allpass filters connected in series can be found in Beigel [2], where the bilinear z-transform is used for determining the phase shift frequency of each stage. Cronin [3] presents a similar implementation for a four-stage phase shifting network. Chamberlin [4] and Smith [5] both discuss using second-order allpass sections for greater control over notch depth, width, and frequency.
Page 58-8
REFERENCES
1. Hartmann, W.M. Flanging and Phasers. Journal of the Audio Engineering Society, Vol. 26, No. 6, pp. 439-443, June 1978. 2. Beigel, Michael I. A Digital Phase Shifter for Musical Applications, Using the Bell Labs (Alles-Fischer) Digital Filter Module. Journal of the Audio Engineering Society, Vol. 27, No. 9, pp. 673-676,September 1979. 3. Cronin, Dennis. Examining Audio DSP Algorithms. Dr. Dobbs Journal, July 1994, p. 78-83. 4. Chamberlin, Hal. Musical Applications of Microprocessors. Second edition. Indianapolis, Indiana: Hayden Books, 1985. 5. Smith, Julius O. An Allpass Approach to Digital Phasing and Flanging. Proceedings of the 1984 ICMC, p. 103-108.
AUTHOR
Page 58-9
Page 58-10
59
SIGNAL MODIFIERS: CONVOLUTION AND MORPHING

convolve
convolve ain, ifilcod, ichannel
59.1
ar1[,ar2] [,ar3][,ar4]
DESCRIPTION
Output is the convolution of signal ain and the impulse response contained in ifilcod. If more than one output signal is supplied, each will be convolved with the same impulse response. Note that it is considerably more efficient to use one instance of the operator when processing a mono input to create stereo, or quad, outputs. Note: this opcode can also be written convle.
ifilcod integer or character-string denoting an impulse response data file. An integer denotes the suffix of a file convolve.m; a character string (in double quotes) gives a filename, optionally a full pathname. If not a fullpath, the file is sought first in the current directory, then in the one given by the environment variable SADIR (if defined). The data file contains the Fourier transform of an impulse response. Memory usage depends on the size of the data file, which is read and held entirely in memory during computation, but which is shared by multiple calls.
PERFORMANCE
convolve implements Fast Convolution. The output of this operator is delayed with respect to the input. The following formulas should be used to calculate the delay: For (1/kr) <= IRdur: Delay = ceil(IRdur * kr) / kr For (1/kr) IRdur: Delay = IRdur * ceil(1/(kr*IRdur)) Where: kr = Csound control rate IRdur = duration, in seconds, of impulse response ceil(n) = smallest integer not smaller than n One should be careful to also take into account the initial delay, if any, of the impulse response. For example, if an impulse response is created from a recording, the soundfile may not have the initial delay included. Thus, one should either ensure that the soundfile has the correct amount of zero padding at the start, or, preferably, compensate for this delay in the orchestra. (the latter method is more efficient). To compensate for the delay in the orchestra, subtract the initial delay from the result calculated using the above formula(s), when calculating the required delay to introduce into the dry audio path. For typical applications, such as reverb, the delay will be in the order of 0.5 to 1.5 seconds, or even longer. This renders the current implementation unsuitable for real time applications. It could conceivably be used for real time filtering however, if the number of taps is small enough.
Signal Modifiers: Convolution and Morphing
Page 59-1
The author intends to create a higher-level operator at some stage, that would mix the wet & dry signals, using the correct amount of delay automatically.
E X A M PL E
Create frequency domain impulse response file: c:\ Csound -Ucvanal l1_44.wav l1_44.cv Determine duration of impulse response. For high accuracy, determine the number of sample frames in the impulse response soundfile, and then compute the duration with: duration = (sample frames)/(sample rate of soundfile) This is due to the fact that the SNDINFO utility only reports the duration to the nearest 10ms. If you have a utility that reports the duration to the required accuracy, then you can simply use the reported value directly. c:\ sndinfo l1_44.wav length = 60822 samples, sample rate = 44100 Duration = 60822/44100 = 1.379s. Determine initial delay, if any, of impulse response. If the impulse response has not had the initial delay removed, then you can skip this step. If it has been removed, then the only way you will know the initial delay is if the information has been provided separately. For this example, lets assume that the initial delay is 60ms. (0.06s) Determine the required delay to apply to the dry signal, to align it with the convolved signal: If kr = 441: 1/kr = 0.0023, which is <= IRdur (1.379s), so: Delay1 = ceil(IRdur * kr) / kr = ceil(608.14) / 441 = 609/441 = 1.38s Accounting for the initial delay: Delay2 = 0.06s Total delay = delay1 - delay2 = 1.38 - 0.06 = 1.32s Create .orc file, e.g.:
; Simple demonstration of CONVOLVE operator, to apply reverb. sr = 44100 kr = 441 ksmps = 100 nchnls = 2 1 instr imix = 0.22 ; Wet/dry mix. Vary as desired. ; NB: Small reverbs often require a much higher ; percentage of wet signal to sound interesting. Large ; reverbs seem require less. Experiment! The wet/dry mix is ; very important - a small change can make a large difference. ivol = 0.9 ; Overall volume level of reverb. May need to adjust ; when wet/dry mix is changed, to avoid clipping. idel = 1.32 ; Required delay to align dry audio with output of convolve. ; This can be automatically calculated within the orc file, ; if desired.
Page 59-2
adry awet1,awet2 adrydel
endin
; input (dry) audio soundin anechoic.wav ; stereo convolved (wet) audio convolve adry,l1_44.cv (1-imix)*adry,idel ; Delay dry signal, to align it with delay ; convolved signal. Apply level ; adjustment here too. ivol*(adrydel+imix*awet1),ivol*(adrydel+imix*awet2) outs ; Mix wet & dry signals, and output
AUTHOR
Greg Sullivan 1996
Page 59-3
59.2
ar
cross2
cross2 ain1, ain2, isize, ioverlap, iwin, kbias
DESCRIPTION
This is an implementation of cross synthesis using FFTs.
This is the size of the FFT to be performed. The larger the size the better the isize frequency response but a sloppy time response. This is the overlap factor of the FFTs, must be a power of two. The best ioverlap settings are 2 and 4. A big overlap takes a long time to compile. This is the ftable that contains the window to be used in the analysis. iwin
PERFORMANCE
ain1 The stimulus sound. Must have high frequencies for best results. ain2 The modulating sound. Must have a moving frequency response (like speech) for best results. kbias The amount of cross synthesis. 1 is the normal, 0 is no cross synthesis.
E X A M PL E S
a1 a2 a3 oscil rand cross2 out 10000, 1, 1 10000 a2, a1, 2048, 4, 2, 1 a3
If ftable one is a speech sound, this will result in speaking white noise. ftable 2 must be a window function (GEN20).
AUTHOR
Page 59-4
60
SIGNAL MODIFIERS: PANNING SPATIALIZATION

pan
pan asig, kx, ky, ifn[, imode[, ioffset]]
AND
60.1
a1, a2, a3, a4
DESCRIPTION
Distribute an audio signal amongst four channels with localization control.
ifn function table number of a stored pattern describing the amplitude growth in a speaker channel as sound moves towards it from an adjacent speaker. Requires extended guard-point. imode (optional) mode of the kx, ky position values. 0 signifies raw index mode, 1 means the inputs are normalized (0 1). The default value is 0. ioffset (optional) offset indicator for kx, ky. 0 infers the origin to be at channel 3 (left rear); 1 requests an axis shift to the quadraphonic center. The default value is 0.
PERFORMANCE
pan takes an input signal asig and distributes it amongst four outputs (essentially quad speakers) according to the controls kx and ky. For normalized input (mode=1) and no offset, the four output locations are in order: left-front at (0,1), right-front at (1,1), leftrear at the origin (0,0), and right-rear at (1,0). In the notation (kx, ky), the coordinates kx and ky, each ranging 0 1, thus control the rightness and forwardness of a sound location. Movement between speakers is by amplitude variation, controlled by the stored function table ifn. As kx goes from 0 to 1, the strength of the right-hand signals will grow from the left-most table value to the right-most, while that of the left-hand signals will progress from the right-most table value to the left-most. For a simple linear pan, the table might contain the linear function 0 1. A more correct pan that maintains constant power would be obtained by storing the first quadrant of a sinusoid. Since pan will scale and truncate kx and ky in simple table lookup, a medium-large table (say 8193) should be used. kx, ky values are not restricted to 0 1. A circular motion passing through all four speakers (inscribed) would have a diameter of root 2, and might be defined by a circle of radius R = root 1/2 with center at (.5,.5). kx, ky would then come from Rcos(angle), Rsin(angle), with an implicit origin at (.5,.5) (i.e. ioffset = 1). Unscaled raw values operate similarly. Sounds can thus be located anywhere in the polar or Cartesian plane; points lying outside the speaker square are projected correctly onto the squares perimeter as for a listener at the center.
Signal Modifiers: Panning and Spatialization
Page 60-1
E X A M PL E
instr k1 phasor k2 tablei k3 tablei a1 oscili a1,a2,a3,a4 pan 1 1/p3 ; fraction of circle k1, 1, 1 ; sin of angle (sinusoid in f1) k1, 1, 1, .25, 1 ; cos of angle (sin offset 1/4 circle) 10000,440, 1 ; audio signal.. a1, k2/2, k3/2, 2, 1, 1 ; sent in a circle (f2=1st quad sin) outq a1, a2, a3, a4 endin
Page 60-2
60.2
a1, a1, a1, a1,
locsig, locsend
locsig locsig locsend locsend asig, kdegree, kdistance, kreverbsend asig, kdegree, kdistance, kreverbsend
a2 a2, a3, a4 a2 a2, a3, a4
DESCRIPTION
locsig takes an input signal and distributes it among 2 or 4 channels using values in degrees to calculate the balance between adjacent channels. It also takes arguments for distance (used to attenuate signals that are to sound as if they are some distance further than the loudspeaker itself), and for the amount the signal that will be sent to reverberators. This unit is based upon the example in the Charles Dodge/Thomas Jerse book, Computer Music, page 320. locsend depends upon the existence of a previously defined locsig. The number of output signals must match the number in the previous locsig. The output signals from locsend are derived from the values given for distance and reverb in the locsig and are ready to be sent to local or global reverb units (see example below). The reverb amount and the balance between the 2 or 4 channels are calculated in the same way as described in the Dodge book (an essential text!).
PERFORMANCE
kdegree value between 0 and 360 for placement of the signal in a 2 or 4 channel space configured as: a1=0, a2=90, a3=180, a4=270 (kdegree=45 would balanced the signal equally between a1 and a2). locsig maps kdegree to sin and cos functions to derive the signal balances (i.e.: asig=1, kdegree=45, a1=a2=.707). kdistance value >= 1 used to attenuate the signal and to calculate reverb level to simulate distance cues. As kdistance gets larger the sound should get softer and somewhat more reverberant (assuming the use of locsend in this case). kreverbsend the percentage of the direct signal that will be factored along with the distance and degree values to derive signal amounts that can be sent to a reverb unit such as reverb, or reverb2.
E X A M PL E
asig some audio signal kdegree kdistance line a1, a2, a3, a4 ar1, ar2, ar3, ar4 ga1 = ga1+ar1 ga2 = ga2+ar2 ga3 = ga3+ar3 ga4 = ga4+ar4 outq instr 99 reverb2 reverb2 reverb2 reverb2 a1, a2, a3, a4 line 0, p3, 360 1, p3, 10 locsig asig, kdegree, kdistance, .1 locsend
endin
a1, a2, a3, a4
a1 a2 a3 a4 outq ga1 = 0 ga2 = 0 ga3 = 0 ga4 = 0
; reverb instrument ga1, 2.5, .5 ga2, 2.5, .5 ga3, 2.5, .5 ga4, 2.5, .5
Page 60-3
In the above example, the signal, asig, is sent around a complete circle once during the duration of a note while at the same time it becomes more and more distant from the listeners location. locsig sends the appropriate amount of the signal internally to locsend. The outputs of the locsend are added to global accumulators in a common Csound style and the global signals are used as inputs to the reverb units in a separate instrument. locsig is useful for quad and stereo panning as well as fixed placed of sounds anywhere between two loudspeakers. Below is an example of the fixed placement of sounds in a stereo field.
instr 1 a1, a2 locsig ar1, ar2 locsend ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 endin instr 99 ; reverb.... endin asig, p4, p5, .1
A few notes:
;place i1 0 1 ;place i1 1 1 ;place i1 2 1 e the sound in the left speaker and near 0 1 the sound in the right speaker and far 90 25 the sound equally between left and right and in the middle ground distance 45 12
The next example shows a simple intuitive use of the distance value to simulate Doppler shift. The same value is used to scale the frequency as is used as the distance input to locsig.
kdistance 1, p3, 10 line kfreq = (ifreq * 340) / (340 + asig oscili kdegree line 0, p3, a1, a2, a3, a4 locsig ar1, ar2, ar3, ar4 locsend kdistance) iamp, kfreq, 1 360 asig, kdegree, kdistance, .1
AUTHOR
Richard Karpen Seattle, Wash 1998 (New in Csound version 3.48)
Page 60-4
60.3
space, spsend, spdist

space spsend spdist asig, ifn, ktime, kreverbsend [,kx, ky] ifn, ktime, [,kx, ky]
a1, a2, a3, a4 a1, a2, a3, a4 k1
DESCRIPTION
space takes an input signal and distributes it among 4 channels using Cartesian xy coordinates to calculate the balance of the outputs. The xy coordinates can be defined in a separate text file and accessed through a Function statement in the score using GEN28, or they can be specified using the optional kx, ky arguments. There advantages to the former are: A graphic user interface can be used to draw and edit the trajectory through the Cartesian plane The file format is in the form time1 X1 Y1 time2 X2 Y2 time3 X3 Y3 allowing the user to define a time-tagged trajectory. space then allows the user to specify a time pointer (much as is used for pvoc, lpread and some other units) to have detailed control over the final speed of movement. spsend depends upon the existence of a previously defined space. The output signals from spsend are derived from the values given for XY and reverb in the space and are ready to be sent to local or global reverb units (see example below). spdist uses the same xy data as space, also either from a text file using GEN28 or from x and y arguments given to the unit directly. The purpose of this unit is to make available the values for distance that are calculated from the xy coordinates. In the case of space the xy values are used to determine a distance which is used to attenuate the signal and prepare it for use in spsend. But it is also useful to have these values for distance available to scale the frequency of the signal before it is sent to the space unit.
PERFORMANCE
The configuration of the XY coordinates in space places the signal in the following way: a1 is -1, 1 a2 is 1, 1 a3 is -1, -1 a4 is 1, -1. This assumes a loudspeaker set up as a1 is left front, a2 is right front, a3 is left back, a4 is right back. Values greater than 1 will result in sounds being attenuated as if in the distance. space considers the speakers to be at a distance of 1; smaller values of XY can be used, but space will not amplify the signal in this case. It will, however balance the signal so that it can sound as if it were within the 4 speaker space. x=0, y=1, will place the signal equally balanced between left and right front channels, x=y=0 will place the signal equally in all 4 channels, and so on. Although there must be 4 output signal from space, it can be used in a 2 channel orchestra. If the XYs are kept so that Y>=1, it should work well to do panning and fixed localization in a stereo field.
Page 60-5
ifn number of the stored function created using GEN28. This function generator reads a text file which contains sets of three values representing the xy coordinates and a time-tag for when the signal should be placed at that location. The file should look like: 0 1 2 2.1 3 5 -1 1 4 -4 10 -40 1 1 4 -4 -10 0
If that file were named move then the GEN28 call in the score would like: f1 0 0 28 move GEN28 takes 0 as the size and automatically allocates memory. It creates values to 10 milliseconds of resolution. So in this case there will be 500 values created by interpolating X1 to X2 to X3 and so on, and Y1 to Y2 to Y3 and so on, over the appropriate number of values that are stored in the function table. In the above example, the sound will begin in the left front, over 1 second it will move to the right front, over another second it move further into the distance but still in the left front, then in just 1/10th of a second it moves to the left rear, a bit distant. Finally over the last .9 seconds the sound will move to the right rear, moderately distant, and it comes to rest between the two left channels (due west!), quite distant. Since the values in the table are accessed through the use of a timepointer in the space unit, the actual timing can be made to follow the files timing exactly or it can be made to go faster or slower through the same trajectory. If you have access to the GUI that allows one to draw and edit the files, there is no need to create the text files manually. But as long as the file is ASCII and in the format shown above, it doesnt matter how it is made! Important: If ifn is 0 then space will take its values for the xy coordinates from kx and ky. ktime index into the table containing the xy coordinates. If used like: ktime line 0, 5, 5 a1, a2, a3, a4 space asig, 1, ktime, ... with the file move described above, the speed of the signals movement will be exactly as described in that file. However: ktime line 0, 10, 5 the signal will move at half the speed specified. Or in the case of: ktime line 5, 15, 0 the signal will move in the reverse direction as specified and 3 times slower! Finally: ktime line 2, 10, 3 will cause the signal to move only from the place specified in line 3 of the text file to the place specified in line 5 of the text file, and it will take 10 seconds to do it. kreverbsend the percentage of the direct signal that will be factored along with the distance as derived from the XY coordinates to calculate signal amounts that can be sent to reverb units such as reverb, or reverb2. kx, ky when ifn is 0, space and spdist will use these values as the XY coordinates to localize the signal. They are optional and both default to 0.
Page 60-6
E X A M PL E
instr 1 asig some audio signal ktime line 0, p3, p10 a1,a2,a3,a4 space asig,1, ktime, .1 ar1,ar2,ar3,ar4 spsend ga1 ga2 ga3 ga4 = = = = ga1+ar1 ga2+ar2 ga3+ar3 ga4+ar4 outq a1, a2, a3, a4 endin instr 99 ; reverb instrument a1 a2 a3 a4 ga1 ga2 ga3 ga4 = = = = 0 0 0 0 reverb2 reverb2 reverb2 reverb2 ga1, ga2, ga3, ga4, outq 2.5, 2.5, 2.5, 2.5, .5 .5 .5 .5
a1, a2, a3, a4
In the above example, the signal, asig, is moved according to the data in Function #1 indexed by ktime. space sends the appropriate amount of the signal internally to spsend. The outputs of the spsend are added to global accumulators in a common Csound style and the global signals are used as inputs to the reverb units in a separate instrument. space can useful for quad and stereo panning as well as fixed placed of sounds anywhere between two loudspeakers. Below is an example of the fixed placement of sounds in a stereo field using XY values from the score instead of a function table.
... a1,a2,a3,a4 ar1,ar2,ar3,ar4 ga1 = ga1+ar1 ga2 = ga2+ar2 outs a1, a2 instr 1 space spsend asig, 0, 0, .1, p4, p5
endin instr 99 ; reverb.... endin
....
A few notes: p4 and p5 are the X and Y values

;place i1 0 1 ;place i1 1 1 ;place i1 2 1 e the sound in the left speaker and near -1 1 the sound in the right speaker and far 45 45 the sound equally between left and right and in the middle ground distance 0 12
Page 60-7
The next example shows a simple intuitive use of the distance values returned by spdist to simulate Doppler shift.
ktime kdist kfreq asig line 0, p3, 10 spdist 1, ktime spdis = (ifreq * 340) / (340 + kdist) iamp, kfreq, 1 oscili space spsend asig, 1, ktime, .1
a1, a2, a3, a4 ar1, ar2, ar3, ar4
The same function and time values are used for both spdist and space. This insures that the distance values used internally in the space unit will be the same as those returned by spdist to give the impression of a Doppler shift!
AUTHOR
Page 60-8
60.4
hrtfer
hrtfer asig, kAz, kElev, HRTFcompact
aLeft, aRight
DESCRIPTION
Output is binaural (headphone) 3D audio.
kAz azimuth value in degrees. Positive values represent position on the right, negative values are positions on the left. kElev elevation value in degrees. Positive values represent position above horizontal, negative values are positions above horizontal. At present, the only file which can be used with hrtfer is HRTFcompact. It must be passed to the opcode as the last argument within quotes as shown above. HRTFcompact my be obtained via anonymous ftp from:
ftp://ftp.maths.bath.ac.uk/pub/dream/utilities/Analysis/HRTFcompact
PERFORMANCE
These unit generators place a mono input signal in a virtual 3D space around the listener by convolving the input with the appropriate HRTF data specified by the opcodes azimuth and elevation values. hrtfer allows these values to be k-values, allowing for dynamic spatialization. hrtfer can only place the input at the requested position because the HRTF is loaded in at i-time (remember that currently, Csound has a limit of 20 files it can hold in memory, otherwise it causes a segmentation fault). The output will need to be scaled either by using balance or by multiplying the output by some scaling constant. Note the sampling rate of the orchestra must be 44.1kHz. This is because 44.1kHz is the sampling rate at which the HRTFs were measured. In order to be used at a different rate, the HRTFs would need to be re-sampled at the desired rate.
E X A M PL E
kaz kel asrc aleft,aright aleftscale arightscale linseg linseg soundin hrtfer = = outs 0, p3, -360 -40, p3, 45 ; move the sound in circle ; around the listener, changing ; elevation as its turning
soundin.1 asrc, kaz, kel, HRTFcompact aleft * 200 aright * 200 aleftscale, arightscale
AUTHORS
Eli Breder & David MacIntyre Montreal 1996
Page 60-9
60.5
vbaplsinit, vbap4, vbap8, vbap16, vbap4move, vbap8move, vbap16move, vbapz, vbapzmove

vbaplsinit idim, ilsnum, idir1, idir2,...
ar1, ar2, ar3, ar4 ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8 ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, ar15, ar16 ar1, ar2, ar3, ar4 ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8 ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, ar15, ar16
vbap4 vbap8 vbap16
asig, iazim, ielev, ispread asig, iazim, ielev, ispread asig, iazim, ielev, ispread
vbap4move vbap8move
asig, ispread, ifldnum, ifld1, ifld2,... asig, ispread, ifldnum, ifld1, ifld2,...
vbap16move asig, ispread, ifldnum, ifld1, ifld2,...
vbapz vbapzmove
inumchnls, istartndx, asig, iazim, ielev, ispread inumchnls, istartndx, ispread, ifldnum, ifld1, ifld2,...
DESCRIPTION
Distribute an audio signal among 2 to 16 output channels or write it to a ZAK array, all with localization control.
idim dimensionality of loudspeaker array. Either 2 or 3. ilsnum number of loudspeakers. In two dimensions, the number can vary from 2 to 16. In three dimensions, the number can vary from 3 and 16. idir1, idir2, etc. directions of loudspeakers. Number of directions must be less than or equal to 16. In two-dimensional loudspeaker positioning, idirn is the azimuth angle respective to nth channel. In three-dimensional loudspeaker positioning, fields are the azimuth and elevation angles of each loudspeaker consequently (azi1, ele1, azi2, ele2, etc.). iazim azimuth angle of the virtual source
Page 60-10
ielev elevation angle of the virtual source ispread spreading of the virtual source (range 0 100). If value is zero, conventional amplitude panning is used. When ispread is increased, the number of loudspeakers used in panning increases. If value is 100, the sound is applied to all loudspeakers. ifldnum number of fields (absolute value must be 2 or larger). If ifldnum is positive, the virtual source movement is a polyline specified by given directions. Each transition is performed in an equal time interval. If ifldnum is negative, specified angular velocities are applied to the virtual source during specified relative time intervals (see below). ifld1, ifld2, etc. azimuth angles or angular velocities, and relative durations of movement phases (see below). inumchnls number of channels to write to the ZA array. Must be in the range 2 256. istartndx first index or position in the ZA array to use
PERFORMANCE
asig audio signal to be panned vbap4, vbap8, and vbap16 take an input signal, asig, and distribute it among 2 to 16 outputs, according to the controls iazim and ielev, and the configured loudspeaker placement. If idim = 2, ielev is set to zero. The distribution is performed using Vector Base Amplitude Panning (VBAP See reference). VBAP distributes the signal using loudspeaker data configured with vbaplsinit. The signal is applied to, at most, two loudspeakers in 2-D loudspeaker configurations, and three loudspeakers in 3-D loudspeaker configurations. If the virtual source is panned outside the region spanned by loudspeakers, the nearest loudspeakers are used in panning. vbap4move, vbap8move, and vbap16move allow the use of moving virtual sources. If ifldnum is positive, the fields represent directions of virtual sources and equal times, iazi1, [iele1,] iazi2, [iele2,], etc. The position of the virtual source is interpolated between directions starting from the first direction and ending at the last. Each interval is interpolated in time that is fraction total_time / number_of_intervals of the duration of the sound event. If ifldnum is negative, the fields represent angular velocities and equal times. The first field is, however, the starting direction, iazi1, [iele1,] iazi_vel1, [iele_vel1,] iazi_vel2, [iele_vel2,] .... Each velocity is applied to the note that is fraction total_time / number_of_velocities of the duration of the sound event. If the elevation of the virtual source becomes greater than 90 degrees or less than 0 degrees, the polarity of angular velocity is changed. Thus the elevational angular velocity produces a virtual source that moves up and down between 0 and 90 degrees. The opcodes vbapz and vbapzmove are the multiple channel analogs of the above opcodes, working an inumchnls and using a ZAK array for output.
E X A M PL E
2-D panning example with stationary virtual sources:
sr kr ksmps nchnls vbaplsinit instr 1 asig oscil a1,a2,a3,a4,a5,a6,a7,a8 = = = = 4100 441 100 4 2, 6,
0, 45, 90, 135, 200, 245, 290, 315
20000, 440, 1 vbap8 asig, p4, 0, 20 ;p4 = azimuth
Page 60-11
;render twice with alternate outq statements ; to obtain two 4 channel .wav files: ; outq outq endin a1,a2,a3,a4 a5,a6,a7,a8
REFERENCE
Ville Pulkki: Virtual Sound Source Positioning Using Vector Base Amplitude Panning Journal of the Audio Engineering Society, 1997 June, Vol. 45/6, p. 456.
AUTHORS
Ville Pulkki Sibelius Academy Computer Music Studio Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Helsinki, Finland May, 2000 (New in Csound version 4.06) John ffitch (vbapz, vbabzmove) University of Bath/Codemist Ltd. Bath, UK May, 2000 (New in Csound version 4.06)
Page 60-12
61
SIGNAL MODIFIERS: SAMPLE LEVEL OPERATORS

samphold, downsamp, upsamp, interp, integ, diff
downsamp upsamp interp integ integ diff diff samphold samphold asig[, iwlen] ksig ksig[, iskip] ksig[, iskip] asig[, iskip] ksig[, iskip] asig[, iskip] xsig, kgate[, ival, ivstor] asig, xgate[, ival, ivstor]
61.1
kr ar ar kr ar kr ar kr ar
DESCRIPTION
Modify a signal by up- or down-sampling, integration, and differentiation.
iwlen (optional) window length in samples over which the audio signal is averaged to determine a downsampled value. Maximum length is ksmps; 0 and 1 imply no window averaging. The default value is 0. iskip (optional) initial disposition of internal save space ( see reson). The default value is 0. ival, ivstor (optional) controls initial disposition of internal save space. If ivstor is zero the internal hold value is set to ival ; else it retains its previous value. Defaults are 0,0 (i.e. init to zero)
PERFORMANCE
downsamp converts an audio signal to a control signal by downsampling. It produces one kval for each audio control period. The optional window invokes a simple averaging process to suppress foldover. upsamp, interp convert a control signal to an audio signal. The first does it by simple repetition of the kval, the second by linear interpolation between successive kvals. upsamp is a slightly more efficient form of the assignment, `asig = ksig. integ, diff perform integration and differentiation on an input control signal or audio signal. Each is the converse of the other, and applying both will reconstruct the original signal. Since these units are special cases of low-pass and high-pass filters, they produce a scaled (and phase shifted) output that is frequency-dependent. Thus diff of a sine produces a cosine, with amplitude 2 * sin(pi * Hz / sr) that of the original (for each component partial); integ will inversely affect the magnitudes of its component inputs. With this understanding, these units can provide useful signal modification.
Signal Modifiers: Sample Level Operators
Page 61-1
samphold performs a sample-and-hold operation on its input according to the value of gate. If gate 0, the input samples are passed to the output; If gate = 0, the last output value is repeated. The controlling gate can be a constant, a control signal, or an audio signal.
E X A M PL E
asrc adif anew agate asamp aout buzz diff balance reson samphold tone 10000,440,20, 1 asrc adif, asrc asrc,0,440 anew, agate asamp,100 ; ; ; ; ; ; band-limited pulse train emphasize the highs but retain the power use a lowpass of the original to gate the new audiosig smooth out the rough edges
Page 61-2
61.2
ir kr ar
ntrpol
ntrpol ntrpol ntrpol isig1, isig2, ipoint [, imin, imax] ksig1, ksig2, kpoint [, imin, imax] asig1, asig2, kpoint [, imin, imax]
DESCRIPTION
Calculates the weighted mean value (i.e. linear interpolation) of two input signals
imin minimum xpoint value (optional, default 0) imax maximum xpoint value (optional, default 1)
PERFORMANCE
xsig1, xsig2 input signals xpoint interpolation point between the two values ntrpol opcode outputs the linear interpolation between two input values. xpoint is the distance of evaluation point from the first value. With the default values of imin and imax, (0 and 1) a zero value indicates no distance from the first value and the maximum distance from the second one. With a 0.5 value, ntrpol will output the mean value of the two inputs, indicating the exact half point between xsig1 and xsig2. A 1 value indicates the maximum distance from the first value and no distance from the second one. The range of xpoint can be also defined with imin and imax to make its management easier. These opcodes are useful for crossfading two signals.
AUTHOR
Gabriel Maldonado Italy October, 1998 (New in Csound version 3.49)
Page 61-3
61.3
ar
fold
fold asig, kincr
DESCRIPTION
Adds artificial foldover to an audio signal.
PERFORMANCE
asig input signal kincr amount of foldover expressed in multiple of sampling rate. Must be >= 1 fold is an opcode which creates artificial foldover. For example, when kincr is equal to 1 with sr=44100, no foldover is added. When kincr is set to 2, the foldover is equivalent to a downsampling to 22050, when it is set to 4, to 11025 etc. Fractional values of kincr are possible, allowing a continuous variation of foldover amount. This can be used for a wide range of special effects.
E X A M PL E
kfreq a1 k1 a1 instr line oscili init fold out endin 1 1,p3,200 10000, 100, 1 8.5 a1, kfreq a1
AUTHOR
Page 61-4
62
ZAK PATCH SYSTEM

The zak opcodes are used to create a system for i-rate, k-rate or a-rate patching. The zak system can be thought of as a global array of variables. These opcodes are useful for performing flexible patching or routing from one instrument to another. The system is similar to a patching matrix on a mixing console or to a modulation matrix on a synthesizer. It is also useful whenever an array of variables is required. The zak system is initialized by the zakinit opcode, which is usually placed just after the other global initializations: sr, kr, ksmps, nchnls. The zakinit opcode defines two areas of memory, one area for i- and k-rate patching, and the other area for a-rate patching. The zakinit opcode may only be called once. Once the zak space is initialized, other zak opcodes can be used to read from, and write to the zak memory space, as well as perform various other tasks.
Zak Patch System
Page 62-1
62.1
zakinit
zakinit isizea, isizek
DESCRIPTION
Establishes zak space. Must be called only once.
isizea the number of audio rate locations for a-rate patching. Each location is actually an array which is ksmps long. isizek the number of locations to reserve for floats in the zk space. These can be written and read at i- and k-rates.
PERFORMANCE
At least one location each is always allocated for both za and zk spaces. There can be thousands or tens of thousands za and zk ranges, but most pieces probably only need a few dozen for patching signals. These patching locations are referred to by number in the other zak opcodes. To run zakinit only once, put it outside any instrument definition, in the orchestra file header, after sr, kr, ksmps, and nchnls.
E X A M PL E
zakinit 10 30 reserves memory for locations 0 to 30 of zk space and for locations 0 to 10 of a-rate za space. With ksmps = 8, this would take 31 floats for zk and 80 floats for za space.
AUTHOR
Zak Patch System
Page 62-2
62.2
ziw, zkw, zaw, ziwm, zkwm, zawm

ziw zkw zaw ziwm zkwm zawm isig, ksig, asig, isig, ksig, asig, indx kndx kndx indx [, imix] kndx [, imix] kndx [, imix]
DESCRIPTION
Write to a location in zk space at either i-rate or k-rate, or a location in za space at a-rate. Writing can be with, or without, mixing.
indx points to the zk location to which to write. isig initializes the value of the zk location.
PERFORMANCE
kndx points to the zk or za location to which to write. ksig value to be written to the zk location. asig value to be written to the za location. ziw writes isig into the zk variable specified by indx. zkw writes ksig into the zk variable specified by kndx. zaw writes asig into the za variable specified by kndx. These opcodes are fast, and always check that the index is within the range of zk or za space. If not, an error is reported, 0 is returned, and no writing takes place. ziwm, zkwm, and zawm are mixing opcodes, i.e. they add the signal to the current value of the variable. If no imix is specified, mixing always occurs, but if imix is specified, imix = 0 will cause overwriting, like ziw, zkw, and zaw, and any other value will cause mixing. Caution: When using the mixing opcodes ziwm, zkwm, and zawm, care must be taken that the variables mixed to, are zeroed at the end (or start) of each k or a cycle. Continuing to add signals to them, can cause their values can drift to astronomical figures. One approach would be to establish certain ranges of zk or za variables to be used for mixing, then use zkcl or zacl to clear those ranges.
E X A M PL E S
instr 1 zkw endin instr 2 zkw endin kzoom, 7 kzoom, p8 ; p8 in the score line determines w where ; in zk space kzoom iswritten
; always writes kzoom to zk location 7
Zak Patch System
Page 62-3
kxxx kdest
instr 3 phasor = zaw endin
1 40+kxxx*16 azoom,kdest
; This will write azoom to ; locations 40 to 55 ; on a one second scan cycle
AUTHOR
Zak Patch System
Page 62-4
62.3
ir kr ar ar
zir, zkr, zar, zarg

zir zkr zar zarg indx kndx kndx kndx, kgain
DESCRIPTION
Read from a location in zk space at i-rate or k-rate, or a location in za space at a-rate.
kndx points to the zk or za location to be read. kgain multiplier for the a-rate signal.
PERFORMANCE
zir reads the signal at indx location in zk space. zkr reads the array of floats at kndx in zk space. zar reads the array of floats at kndx in za space, which are ksmps number of a-rate floats to be processed in a k cycle. zarg is similar to zar, but multiplies the a-rate signal by a k-rate value kgain.
AUTHOR
Zak Patch System
Page 62-5
62.4
kr ar
zkmod, zamod, zkcl,

zkmod zamod zkcl zacl ksig, kzkmod asig, kzamod kfirst, klast kfirst, klast
zacl
DESCRIPTION
Clear and modulate the za and zk spaces.
PERFORMANCE
ksig the input signal kzkmod controls which zk variable is used for modulation. A positive value means additive modulation, a negative value means multiplicative modulation. A value of 0 means no change to ksig. kzkmod can be i-rate or k-rate kfirst first zk or za location in the range to clear. klast last zk or za location in the range to clear. zkmod facilitates the modulation of one signal by another, where the modulating signal comes from a zk variable. Either additive or multiplicative modulation can be specified. zamod modulates one a-rate signal by a second one, which comes from a za variable. The location of the modulating variable is controlled by the i-rate or k-rate variable kzamod. This is the a-rate version of zkmod zkcl clears one or more variables in the zk space. This is useful for those variables which are used as accumulators for mixing k-rate signals at each cycle, but which must be cleared before the next set of calculations. zacl clears one or more variables in the za space. This is useful for those variables which are used as accumulators for mixing a-rate signals at each cycle, but which must be cleared before the next set of calculations.
E X A M PL E S
k1 a1 zkmod zamod ksig, 23 asig, -402 ; adds value at location 23 to ksig ; multiplies asig by value at location 402
AUTHOR
Zak Patch System
Page 62-6
63
OPERATIONS USING SPECTRAL DATA TYPES

These units generate and process non-standard signal data types, such as down-sampled time-domain control signals and audio signals, and their frequency-domain (spectral) representations. The new data types (d-, w-) are self-defining, and the contents are not processable by any other Csound units. These unit generators are experimental, and subject to change between releases; they will also be joined by others later.
Operations Using Spectral Data Types
Page 63-1
63.1
specaddm, specdiff, specscal, spechist, specfilt

specaddm specdiff specscal spechist specfilt wsig1, wsig2[, imul2] wsigin wsigin, ifscale, ifthresh wsigin wsigin, ifhtim
wsig wsig wsig wsig wsig
imul2 (optional) if non-zero, scale the wsig2 magnitudes before adding. The default value is 0.
PERFORMANCE
specaddm do a weighted add of two input spectra. For each channel of the two input spectra, the two magnitudes are combined and written to the output according to: magout = mag1in + mag2in * imul2. The operation is performed whenever the input wsig1 is sensed to be new. This unit will (at Initialization) verify the consistency of the two spectra (equal size, equal period, equal mag types). specdiff find the positive difference values between consecutive spectral frames. At each new frame of wsigin, each magnitude value is compared with its predecessor, and the positive changes written to the output spectrum. This unit is useful as an energy onset detector. specscal scale an input spectral datablock with spectral envelopes. Function tables ifthresh and ifscale are initially sampled across the (logarithmic) frequency space of the input spectrum; then each time a new input spectrum is sensed the sampled values are used to scale each of its magnitude channels as follows: if ifthresh is non-zero, each magnitude is reduced by its corresponding table-value (to not less than zero); then each magnitude is rescaled by the corresponding ifscale value, and the resulting spectrum written to wsig. spechist accumulate the values of successive spectral frames. At each new frame of wsigin, the accumulations-to-date in each magnitude track are written to the output spectrum. This unit thus provides a running histogram of spectral distribution. specfilt filter each channel of an input spectrum. At each new frame of wsigin, each magnitude value is injected into a 1st-order lowpass recursive filter, whose half-time constant has been initially set by sampling the ftable ifhtim across the (logarithmic) frequency space of the input spectrum. This unit effectively applies a persistence factor to the data occurring in each spectral channel, and is useful for simulating the energy integration that occurs during auditory perception. It may also be used as a timeattenuated running histogram of the spectral distribution.
E X A M PL E
wsig2 wsig3 specdisp specdisp specdiff wsig1 specfilt specfil wsig2, 2 wsig2, wsig2 .1 wsig3, wsig3 .1 ; sense onsets ; absorb slowly ; & display both spectra
Page 63-2
63.2
koct, kamp
specptrk
specptrk wsig, kvar, ilo, ihi, istr, idbthresh, inptls,// irolloff[, iodd, iconfs, interp, ifprd, iwtflg]
DESCRIPTION
Estimate the pitch of the most prominent complex tone in the spectrum.
ilo, ihi, istr pitch range conditioners (low, high, and starting) expressed in decimal octave form. idbthresh energy threshold (in decibels) for pitch tracking to occur. Once begun, tracking will be continuous until the energy falls below one half the threshold (6 dB down), whence the koct and kamp outputs will be zero until the full threshold is again surpassed. idbthresh is a guiding value. At initialization it is first converted to the idbout mode of the source spectrum (and the 6 dB down point becomes .5, .25, or 1/root 2 for modes 0, 2 and 3). The values are also further scaled to allow for the weighted partial summation used during correlation.The actual thresholding is done using the internal weighted and summed kamp value that is visible as the second output parameter. inptls, irolloff number of harmonic partials used as a matching template in the spectrally-based pitch detection, and an amplitude rolloff for the set expressed as some fraction per octave (linear, so dont roll off to negative). Since the partials and rolloff fraction can affect the pitch following, some experimentation will be useful: try 4 or 5 partials with .6 rolloff as an initial setting; raise to 10 or 12 partials with rolloff .75 for complex timbres like the bassoon (weak fundamental). Computation time is dependent on the number of partials sought. The maximum number is 16. iodd (optional) if non-zero, employ only odd partials in the above set (e.g. inptls of 4 would employ partials 1,3,5,7). This improves the tracking of some instruments like the clarinet The default value is 0 (employ all partials). iconfs (optional) number of confirmations required for the pitch tracker to jump an octave, pro-rated for fractions of an octave (i.e. the value 12 implies a semitone change needs 1 confirmation (two hits) at the spectrum generating iprd). This parameter limits spurious pitch analyses such as octave errors. A value of 0 means no confirmations required; the default value is 10. interp (optional) if non-zero, interpolate each output signal (koct, kamp) between incoming wsig frames. The default value is 0 (repeat the signal values between frames). ifprd (optional) if non-zero, display the internally computed spectrum of candidate fundamentals. The default value is 0 (no display). iwtftg (optional) wait flag. If non-zero, hold each display until released by the user. The default value is 0 (no wait).
PERFORMANCE
At note initialization this unit creates a template of inptls harmonically related partials (odd partials, if iodd non-zero) with amplitude rolloff to the fraction irolloff per octave. At each new frame of wsig, the spectrum is cross-correlated with this template to provide an internal spectrum of candidate fundamentals (optionally displayed). A likely pitch/amp pair (koct, kamp, in decimal octave and summed idbout form) is then estimated. koct varies from the previous koct by no more than plus or minus kvar decimal octave units. It is also guaranteed to lie within the hard limit range ilo ihi (decimal octave low and high
The Public Csound Reference Manual Version 4.10 Operations Using Spectral Data Types Page 63-3
pitch). kvar can be dynamic, e.g. onset amp dependent. Pitch resolution uses the originating spectrum ifrqs bins/octave, with further parabolic interpolation between adjacent bins. Settings of root magnitude, ifrqs = 24, iq = 15 should capture all the inflections of interest. Between frames, the output is either repeated or interpolated at the k-rate. (See spectrum.)
E X A M PL E
a1,a2 krms kvar wsig ins ; read a stereo clarinet input a1, 20 ; find a monaural rms value rms = 0.6 + krms/8000 ; & use to gate the pitch variance spectrum a1, .01, 7, 24, 15, 0, 3 ; get a 7-oct spectrum, 24 bibs/oct specdisp wsig, .2 ; display this and now estimate koct,ka spectrk wsig, kvar, 7.0, 10, 9, 20, 4, .7, 1, 5, 1, .2 ; the ; pch and amp aosc ka*ka*10, cpsoct(koct),2 ; & generate \ new tone with these oscil koct = (koct<7.0?7.0:koct) ; replace non pitch with low C ; & display the pitch track display koct-7.0, .25, 20 ; plus the summed root mag display ka, .25, 20 a1, aosc ; output 1 original and 1 new outs ; track
Page 63-4
63.3
ksum
specsum, specdisp
specsum specdisp wsig[, interp] wsig, iprd[, iwtflg]
interp (optional) if non-zero, interpolate the output signal (koct or ksum). The default value is 0 (repeat the signal value between changes). iwtflg (optional) wait flag. If non-zero, hold each display until released by the user. The default value is 0 (no wait).
PERFORMANCE
specsum sum the magnitudes across all channels of the spectrum. At each new frame of wsig, the magnitudes are summed and released as a scalar ksum signal. Between frames, the output is either repeated or interpolated at the k-rate. This unit produces a k-signal summation of the magnitudes present in the spectral data, and is thereby a running measure of its moment-to-moment overall strength. specdisp display the magnitude values of spectrum wsig every iprd seconds (rounded to some integral number of wsigs originating iprd).
E X A M PL E
ksum specsum if koct specptrk kgoto zero: koct contin: . wsig, 1 ksum < 2000 wsig contin = 0 ; sum the spec bins, and ksmooth kgoto zero ; if sufficient amplitude ; pitch-track the signal ; else output zero
Page 63-5
63.4
wsig
spectrum
spectrum xsig, iprd, iocts, ifrqs, iq[,ihann, idbout, idsprd, idsinrs]
DESCRIPTION
Generate a constant-Q, exponentially-spaced DFT across all octaves of a multiplydownsampled control or audio input signal.
ihann (optional) apply a Hamming or Hanning window to the input. The default is 0 (Hamming window) idbout (optional) coded conversion of the DFT output: 0 = magnitude, 1 = dB, 2 = mag squared, 3 = root magnitude. The default value is 0 (magnitude). idisprd (optional) if non-zero, display the composite downsampling buffer every idisprd seconds. The default value is 0 (no display). idsines (optional) if non-zero, display the Hamming or Hanning windowed sinusoids used in DFT filtering. The default value is 0 (no sinusoid display).
PERFORMANCE
This unit first puts signal asig or ksig through iocts of successive octave decimation and downsampling, and preserves a buffer of down-sampled values in each octave (optionally displayed as a composite buffer every idisprd seconds). Then at every iprd seconds, the preserved samples are passed through a filter bank (ifrqs parallel filters per octave, exponentially spaced, with frequency/bandwidth Q of iq), and the output magnitudes optionally converted (idbout ) to produce a band-limited spectrum that can be read by other units. The stages in this process are computationally intensive, and computation time varies directly with iocts, ifrqs, iq, and inversely with iprd. Settings of ifrqs = 12, iq = 10, idbout = 3, and iprd = .02 will normally be adequate, but experimentation is encouraged. ifrqs currently has a maximum of 120 divisions per octave. For audio input, the frequency bins are tuned to coincide with A440. This unit produces a self-defining spectral datablock wsig, whose characteristics used (iprd, iocts, ifrqs, idbout) are passed via the data block itself to all derivative wsigs. There can be any number of spectrum units in an instrument or orchestra, but all wsig names must be unique.
E X A M PL E
asig in ; get external audio wsig asig,.02,6,12,33,0,1,1 ; downsample in 6 octs & calc spectrum a 72 pt dft (Q 33, dB out) every 20 msecs
Page 63-6
64
64.1
SIGNAL INPUT
AND
OUTPUT: INPUT
in, ins, inq, inh, ino, soundin, diskin
ar1 in ar1, ar2 ins ar1, ar2, inq ar3, ar4 ar1, ar2, inh ar3, ar4, ar5, ar6 ar1, ar2, ino ar3, ar4, ar5, ar6, ar7, ar8 ar1 soundin ar1, ar2 soundin ar1, ar2, soundin ar3, ar4 ar1[,ar2 diskin [,a3,ar4]]
ifilcod[, iskptim[, iformat]] ifilcod[, iskptim[, iformat]] ifilcod[, iskptim[, iformat]] ifilcod, kpitch[, iskiptim [,iwraparound[, iformat]]]
DESCRIPTION
These units read audio data from an external device or stream.
ifilcod integer or character-string denoting the source soundfile name. An integer denotes the file soundin.filcod ; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is sought first in the current directory, then in that given by the environment variable SSDIR (if defined) then by SFDIR. See also GEN01. iskptim (optional) time in seconds of input sound to be skipped. The default value is 0. iformat (optional) specifies the audio data file format: 1 = 8-bit signed char (high-order 8 bits of a 16-bit integer), 2 = 8-bit A-law bytes, 3 = 8-bit U-law bytes, 4 = 16-bit short integers, 5 = 32-bit long integers, 6 = 32-bit floats. If iformat = 0 it is taken from the soundfile header, and if no header from the Csound -o command flag. The default value is 0. iwraparound 1=on, 0=off (wraps around to end of file either direction) kpitch can be any real number. a negative number signifies backwards playback. The given number is a pitch ratio, where:
Signal Input and Output: Input
Page 64-1
1 = 2 = 3 = .5 = .25 = -1 = -2 =
norm pitch, oct higher, 12th higher, etc; oct lower, 2oct lower, etc; norm pitch backwards, oct higher backwards, etc..
PERFORMANCE
in, ins, inq, inh, ino copy the current values from the standard audio input buffer. If the command-line flag -i is set, sound is read continuously from the audio input stream (e.g. stain or a soundfile) into an internal buffer. Any number of these units can read freely from this buffer. soundin is functionally an audio generator that derives its signal from a pre-existing file. The number of channels read in is controlled by the number of result cells, a1, a2, etc., which must match that of the input file. A soundin unit opens this file whenever the host instrument is initialized, then closes it again each time the instrument is turned off. There can be any number of soundin units within a single instrument or orchestra; also, two or more of them can read simultaneously from the same external file. diskin is identical to soundin, except that it can alter the pitch of the sound that is being read.
AUTHORS
Barry Vercoe, Matt Ingols/Mike Berry MIT, Mills College 1993-1997
Page 64-2
64.2
inx, in32, inch, inz

inx
ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, ar9, ar10, ar11, ar12, ar13, ar14, ar15, ar16
ar1, ar2, ar3, ar4, ar5, ar6, ar7, ar8, in32 ar9, ar10, ar11, ar12, ar13, ar14, ar15, ar16, ar17, ar18, ar19, ar20, ar21, ar22, ar23, ar24, ar25, ar26, ar27, ar28, ar29, ar30, ar31, ar32 ar1 inch inz ksig1 ksig1
DESCRIPTION
These units read multi-channel audio data from an external device or stream.
PERFORMANCE
inx and in32 read 16 and 32 channel inputs, respectively. inch reads from a numbered channel determined by ksig1 into a1. inz reads audio samples in nchnls into a ZAK array starting at ksig1.
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK May, 2000 (New in Csound version 4.06)
Page 64-3
Page 64-4
65
65.1
SIGNAL INPUT
AND
OUTPUT: OUTPUT
soundout, soundouts, out, outs1, outs2, outs, outq1, outq2, outq3, outq4, outq, outh, outo
soundout soundouts out outs1 outs2 outs outq1 outq2 outq3 outq4 outq outh outo asig, ifilcod[, iskptim] asig, ifilcod[, iskptim] asig asig asig asig1, asig2 asig asig asig asig asig1, asig2, asig3, asig4 asig1, asig2, asig3, asig4, asig5, asig6 asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8
DESCRIPTION
These units write audio data to an external device or stream.
ifilcod integer or character-string denoting the destination soundfile name. An integer denotes the file soundin.filcod; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is sought first in the current directory, then in that given by the environment variable SSDIR (if defined) then by SFDIR. See also GEN01. iskptim (optional) time in seconds of input sound to be skipped. The default value is 0.
PERFORMANCE
out, outs, outq, outh, outo send audio samples to an accumulating output buffer (created at the beginning of performance) which serves to collect the output of all active instruments before the sound is written to disk. There can be any number of these output units in an instrument. The type (mono, stereo, quad, hex, or oct) should agree with nchnls, but as of version 3.50, will attempt to change and incorrect opcode, to arguer with nchnls statement. Units can be chosen to direct sound to any particular channel: outs1 sends to stereo channel 1, outq3 to quad channel 3, etc. soundout and soundouts write audio output to a disk file. soundouts is currently not implemented.
Signal Input and Output: Output
Page 65-1
AUTHORS
Barry Vercoe, Matt Ingols/Mike Berry MIT, Mills College 1993-1997
Page 65-2
65.2
outx, out32, outc, outch, outz

outx asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8, asig9, asig10, asig11, asig12, asig13, asig14, asig15, asig16 asig1, asig2, asig3, asig4, asig5, asig6, asig7, asig8, asig10, asig11, asig12, asig13, asig14, asig15, asig16, asig17, asig18, asig19, asig20, asig21, asig22, asig23, asig24, asig25, asig26, asig27, asig28, asig29, asig30, asig31, asig32 asig1[, asig2,...] ksig1, asig1, ksig2, asig2, ... ksig1
out32
outc outch outz
DESCRIPTION
These units write multi-channel audio data to an external device or stream.
PERFORMANCE
outx and out32 output 16 and 32 channels of audio. outc outputs as many channels as provided. Any channels greater than nchnls are ignored, and zeros are added as necessary outch outputs asig1 on the channel determined by ksig1, asig2 on the channel determined by ksig2, etc. outz outputs from a ZAK array, for nchnls of audio.
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK May, 2000 (New in Csound version 4.06)
Page 65-3
Page 65-4
66
SIGNAL INPUT F I L E I/O
AND
OUTPUT:
66.1
dumpk, dumpk2, dumpk3, dumpk4, readk, readk2, readk3, readk4

ksig, ifilname, iformat, iprd ksig1, ksig2, ifilname, iformat, iprd ksig1, ksig2, ksig3, ifilname, iformat, iprd ksig1, ksig2, ksig3, ksig4, ifilname, iformat, iprd ifilname, iformat, iprd[, ipol] ifilname, iformat, iprd[, ipol] ifilname, iformat, iprd[, ipol] ifilname, iformat, iprd[, ipol]
dumpk dumpk2 dumpk3 dumpk4 kr1 readk kr1,kr2 readk2 kr1,kr2, readk3 kr3 kr1,kr2, readk4 kr3,kr4
DESCRIPTION
Periodically write orchestra control-signal values to a named external file in a specific format.
ifilname character string(in double quotes, spaces permitted ) denoting the external file name. May either be a full path name with target directory specified or a simple filename to be created within the current directory iformat specifies the output data format: 1 4 5 6 8 = = = = = 8-bit signed char(high order 8 bits of a 16-bit integer 16-bit short integers 32bit long integers 32-bit floats, 7=ASCII long integers ASCII floats (2 decimal places)
Note that A-law and U-law output are not available, and that all formats except the last two are binary. The output file contains no header information. iprd the period of ksig output i seconds, rounded to the nearest orchestra control period. A value of 0 implies one control period (the enforced minimum), which will create an output file sampled at the orchestra control rate. ipol (optional) if non-zero, and iprd implies more than one control period, interpolate the k- signals between the periodic reads from the external file. The default value is 0 (repeat each signal between frames). Currently not supported.
The Public Csound Reference Manual Version 4.10 File I/O Page 66-1
Signal Input and Output:
PERFORMANCE
These units allow up to four generated control signal values to be read or saved in a named external file. The file contains no self-defining header information, but is a regularly sampled time series, suitable for later input or analysis. There may be any number of readk units in an instrument or orchestra, and they may read from the same or different files. There may be any number of dumpk units in an instrument or orchestra, but each must write to a different file.
O PCO D E H I S T O R Y
dumpk opcodes were originally called kdump. As of Csound version 3.493 that name is deprecated. dumpk should be used instead of kdump. The readk opcodes were originally called kread, but were not implemented until Csound version 3.52. However, the optional readk argument, ipol is ignored. This situation is expected to be corrected in a later release.
E X A M PL E
knum = ktemp tempest koct knum+1 ; at each k-period krms, .02, .1, 3, 2, 800, .005, 0, 60, 4, .1, .995 ;estimate the tempo specptrk wsig, 6, .9, 0 ;and the pitch knum, ktemp, cpsoct(koct), what happened when, 8 0 dumpk3 ;& save them
66.2
fout, foutk, fouti, foutir, fiopen

fout foutk fouti foutir ifilename, iformat, aout1[, aout2, aout3,...,aoutN] ifilename, iformat, kout1[, kout2, kout3,....,koutN] ihandle, iformat, iflag, iout1[, iout2, iout3,....,ioutN] ihandle, iformat, iflag, iout1[, iout2,\\ iout3,....,ioutN] ifilename,imode
ihandle
fiopen
DESCRIPTION
fout, foutk, fouti and foutir output N a-, k-, or i-rate signals to a specified file of N channels. fiopen can be used to open a file in one of the specified modes.
ifilename a double-quote delimited string file name iformat a flag to choose output file format: for fout and foutk only: 0 32-bit floating point samples without header (binary PCM multichannel file) 1 16-bit integers without header (binary PCM multichannel file) 2 16-bit integers with .wav type header (Microsoft WAV mono or stereo file) for fouti and foutir only: 0 floating point in text format 1 32-bit floating point in binary format
iflag choose the mode of writing to the ASCII file (valid only in ASCII mode; in binary mode iflag has no meaning, but it must be present anyway). iflag can be a value chosen among the following: 0 line of text without instrument prefix 1 line of text with instrument prefix (see below) 2 reset the time of instrument prefixes to zero (to be used only in some particular cases. See below)
iout,..., ioutN values to be written to the file imode choose the mode of opening the file. imode can be a value chosen among the following: 0 1 2 3 open a text file for writing open a text file for reading open a binary file for writing open a binary file for reading
PERFORMANCE
aout1,... aoutN signals to be written to the file kout1,...koutN signals to be written to the file fout (file output) writes samples of audio signals to a file with any number of channels. Channel number depends by the number of aoutN variables (i.e. a mono signal with only an a-rate argument, a stereo signal with two a-rate arguments etc.) Maximum number of channels is fixed to 64. Multiple fout opcodes can be present in the same instrument, referring to different files.
Notice that, unlike out, outs and outq, fout does not zero the audio variable, so you must zero it after calling fout, if polyphony is to be used. You can use incr and clear opcodes for this task. foutk operates in the same way as fout, but with k-rate signals. iformat can be set only to 0 or 1. fouti and foutir write i-rate values to a file. The main use of these opcodes is to generate a score file during a real-time session. For this purpose, the user should set iformat to 0 (text file output) and iflag to 1, which enable the output of a prefix consisting of the strings inum, actiontime, and duration, before the values of iout1...ioutN arguments. The arguments in the prefix refer to instrument number, action time and duration of current note. The difference between fouti and foutir is that, in the case of fouti, when iflag is set to 1, the duration of the first opcode is undefined (so it is replaced by a dot). Whereas, foutir is defined at the end of note, so the corresponding text line is written only at the end of the current note (in order to recognize its duration). The corresponding file is linked by the ihandle value generated by the fiopen opcode (see below). So fouti and foutir can be used to generate a Csound score while playing a real-time session. fiopen opens a file to be used by the fout family of opcodes. It must be defined in the header section, external to any instruments. It returns a number, ihandle, which is unequivocally referring to the opened file. Notice that fout and foutk can use either a string containing a file pathname, or a handlenumber generated by fiopen. Whereas, with fouti and foutir, the target file can be only specified by means of a handle-number.
AUTHOR
66.3
fin, fink, fini

fin fink fini ifilename, iskipframes, iformat, ain1[, ain2,\\ ain3,...,ainN] ifilename, iskipframes, iformat, kin1[, kin2,\\ kin3,...,kinN] ifilename, iskipframes, iformat, in1[, in2,\\ in3,...,inN]
DESCRIPTION
Read signals from a file (at a-, k-, and i-rate)
ifilename input file name (can be a string or a handle number generated by fiopen) iskipframes number of frames to skip at the start (every frame contains a sample of each channel) iformat a number specifying the input file format for fin and fink: 0 32 bit floating points without header 1 16 bit integers without header and for fini: 0 floating points in text format (loop; see below) 1 floating points in text format (no loop; see below) 2 32 bit floating points in binary format (no loop)
PERFORMANCE
fin (file input) is the complement of fout: it reads a multichannel file to generate audio rate signals. At the present time no header is supported for the file format. The user must be sure that the number of channels of the input file is the same as the number of ainX arguments. fink is the same as fin, but operates at k-rate. fini is the complement of fouti and foutir, it reads the values each time the corresponding instrument note is activated. When iformat is set to 0, if the end of file is reached, the file pointer is zeroed, restarting the scan from the beginning. When iformat is set to 1 or 2, no looping is enabled, so at the end of file, the corresponding variables will be filled with zeroes.
AUTHOR
66.4
vincr, clear
vincr clear asig, aincr avar1[,avar2, avar3,...,avarN]
DESCRIPTION
vincr increments an audio variable of another signal, i.e. accumulates output. clear zeroes a list of audio signals.
PERFORMANCE
asig audio variable to be incremented aincr incrementing signal avar1 [,avar2, avar3,...,avarN] signals to be zeroed vincr (variable increment) and clear are intended to be used together. vincr stores the result of the sum of two audio variables into the first variable itself (which is intended to be used as an accumulator in polyphony). The accumulator variable can be used for output signal by means of fout opcode. After the disk writing operation, the accumulator variable should be set to zero by means of clear opcode (or it will explode).
AUTHOR
67
SIGNAL INPUT FILE QUERIES
AND
OUTPUT: SOUND
67.1
ir ir ir ir
filelen, filesr, filenchnls, filepeak

filelen filesr filenchnls filepeak ifilcod ifilcod ifilcod ifilcod[, ichnl]
DESCRIPTION
Obtains information about a sound file.
ifilecod sound file to be queried ichnl channel to be used in calculating the peak value. Default is 0. ichnl = 0 returns peak value of all channels ichnl > 0 returns peak value of ichnl
PERFORMANCE
filelen returns the length of the sound file ifilcod in seconds. filesr returns the sample rate of the sound file ifilcod. filenchnls returns the number of channels in the sound file ifilcod. filepeak returns the peak absolute value of the sound file ifilcod. Currently, filepeak supports only AIFF-C float files.
AUTHOR
Matt Ingalls July, 1999 New in Csound version 3.57
Signal Input and Output: Sound File Queries
Page 67-1
Page 67-2
68
SIGNAL INPUT AND OUTPUT: PRINTING AND DISPLAY

print, display, dispfft
print display dispfft iarg[, iarg,...] xsig, iprd[, inprds[, iwtflg]] xsig, iprd, iwsiz[, iwtyp[, idbouti[, iwtflg]]]
68.1
DESCRIPTION
These units will print orchestra init-values, or produce graphic display of orchestra control signals and audio signals. Uses X11 windows if enabled, else (or if -g flag is set) displays are approximated in ASCII characters.
iprd the period of display in seconds. iwsiz size of the input window in samples. A window of iwsiz points will produce a Fourier transform of iwsiz/2 points, spread linearly in frequency from 0 to sr/2. iwsiz must be a power of 2, with a minimum of 16 and a maximum of 4096. The windows are permitted to overlap. iwtyp (optional) window type. 0 = rectangular, 1 = Hanning. The default value is 0 (rectangular). idbout (optional) units of output for the Fourier coefficients. 0 = magnitude, 1 = decibels. The default is 0 (magnitude). iwtflg (optional) wait flag. If non-zero, each display is held until released by the user. The default value is 0 (no wait).
PERFORMANCE
print print the current value of the i-time arguments (or expressions) iarg at every i-pass through the instrument. display display the audio or control signal xsig every iprd seconds, as an amplitude vs. time graph. dispfft display the Fourier Transform of an audio or control signal (asig or ksig) every iprd seconds using the Fast Fourier Transform method.
E X A M PL E
k1 envlpx display l, .03, p3, .05, l, .5, .0l ; generate a note envelope k1, p3 ; and display entire shape
Signal Input and Output: Printing and Display
Page 68-1
68.2
printk, printks
printk printks itime, kval, [ispace] txtstring, itime, kval1, kval2, kval3, kval4
DESCRIPTION
These opcodes are intended to facilitate the debugging of orchestra code.
itime time in seconds between printings. (Default 1 second.) ispace (optional) number of spaces to insert before printing. (Max 130.) txtstring text to be printed. Can be up to 130 characters and must be in double quotes.
PERFORMANCE
kvalx The k-rate values to be printed. These are specified in txtstring with the standard C value specifier %f, in the order given. Use 0 for those which are not used. printk prints one k-rate value on every k cycle, every second or at intervals specified. First the instrument number is printed, then the absolute time in seconds, then a specified number of spaces, then the kval value. The variable number of spaces enables different values to be spaced out across the screen so they are easier to view. printks prints numbers and text, with up to four printable numbers which can be i- or krate values. printks is highly flexible, and if used together with cursor positioning codes, could be used to write specific values to locations in the screen as the Csound processing proceeds. A special mode of operation allows this printks to convert kval1 input parameter into a 0 to 255 value and to use it as the first character to be printed. This enables a Csound program to send arbitrary characters to the console. To achieve this, make the first character of the string a # and then, if desired continue with normal text and format specifiers. Three more format specifies may be used they access kval2, kval3 and kval4. Both these opcodes can be run on every k cycle they are run in the instrument. To every accomplish this, set itime to 0. When itime is not 0, the opcode print on the first k cycle it is called, and subsequently when every itime period has elapsed. The time cycles start from the time the opcode is initialized typically the initialization of the instrument.
P R I N T O U T PU T F O R M A T T I N G
Standard C language printf() control characters may be used, but must be prefaced with an additional backslash: \\n or \\N \\t or \\T Newline Tab
Page 68-2
The standard C language %f format is used to print kval1, kval2, kval3, and kval4. For example: %f %6.2f %5.0p prints with full precision: 123.456789 prints 1234.56 prints 12345
E X A M PL E S
The following:
printks \Volume = %6.2f Freq = %8.3f\n\, 0.1, kvol, kfreq, 0, 0 would print: Volume = 1234.56 Freq = 12345.678
The following:
printks \#x\\y = %6.2\n\, 0.1, kxy, 0, 0, 0 would print a tab character followed by: x\y = 1234.56
AUTHOR
Page 68-3
68.3
printk2
printk2 kvar [, numspaces]
numspaces number of space characters printed before the value of kvar
PERFORMANCE
kvar signal to be printed Derived from Robin Whittles printk, prints a new value of kvar each time kvar changes. Useful for monitoring MIDI control changes when using sliders. WARNING! dont use this opcode with normal, continuously variant k-signals, because it can hang the computer, as the rate of printing is too fast.
AUTHOR
Page 68-4
69
69.1
THE STANDARD NUMERIC SCORE

Preprocessing of Standard Scores
A Score (a collection of score statements) is divided into time-ordered sections by the s statement. Before being read by the orchestra, a score is preprocessed one section at a time. Each section is normally processed by 3 routines: Carry, Tempo, and Sort.
CARRY
Within a group of consecutive i statements whose p1 whole numbers correspond, any pfield left empty will take its value from the same pfield of the preceding statement. An empty pfield can be denoted by a single point (.) delimited by spaces. No point is required after the last nonempty pfield. The output of Carry preprocessing will show the carried values explicitly. The Carry Feature is not affected by intervening comments or blank lines; it is turned off only by a non-i statement or by an i statement with unlike p1 whole number. Three additional features are available for p2 alone: +, ^ + x, and ^ x. The symbol + in p2 will be given the value of p2 + p3 from the preceding i statement. This enables note action times to be automatically determined from the sum of preceding durations. The + symbol can itself be carried. It is legal only in p2. E.g.: the statements i1 i . I 0 + .5 100
will result in i1 i1 i1 0 .5 1 .5 .5 .5 100 100 100
The symbols ^ + x and ^ x determine the current p2 by adding or subtracting, respectively, the value of x from the preceding p2. These may be used in p2 only. The Carry feature should be used liberally. Its use, especially in large scores, can greatly reduce input typing and will simplify later changes.
T EM PO
This operation time warps a score section according to the information in a t statement. The tempo operation converts p2 (and, for i statements, p3) from original beats into real seconds, since those are the units required by the orchestra. After time warping, score files will be seen to have orchestra-readable format demonstrated by the following: i p1 p2beats p2seconds p3beats p3seconds p4 p5 ....
The Standard Numeric Score
Page 69-1
SORT
This routine sorts all action-time statements into chronological order by p2 value. It also sorts coincident events into precedence order. Whenever an f statement and an i statement have the same p2 value, the f statement will precede. Whenever two or more i statements have the same p2 value, they will be sorted into ascending p1 value order. If they also have the same p1 value, they will be sorted into ascending p3 value order. Score sorting is done section by section (see s statement). Automatic sorting implies that score statements may appear in any order within a section.
NOTE
The operations Carry, Tempo and Sort are combined in a 3-phase single pass over a score file, to produce a new file in orchestra-readable format ( see the Tempo example). Processing can be invoked either explicitly by the Scsort command, or implicitly by Csound which processes the score before calling the orchestra. Source-format files and orchestrareadable files are both in ASCII character form, and may be either perused or further modified by standard text editors. User-written routines can be used to modify score files before or after the above processes, provided the final orchestra-readable statement format is not violated. Sections of different formats can be sequentially batched; and sections of like format can be merged for automatic sorting.
Page 69-2
69.2
Next-P and Previous-P Symbols

At the close of any of the operations Carry, Tempo, and Sort, three additional score features are interpreted during file writeout: next-p, previous-p, and ramping. i statement pfields containing the symbols npx or ppx (where x is some integer) will be replaced by the appropriate pfield value found on the next i statement (or previous i statement) that has the same p1. For example, the symbol np7 will be replaced by the value found in p7 of the next note that is to be played by this instrument. np and pp symbols are recursive and can reference other np and pp symbols which can reference others, etc. References must eventually terminate in a real number or a ramp symbol. Closed loop references should be avoided. np and pp symbols are illegal in p1,p2 and p3 (although they may reference these). np and pp symbols may be Carried. np and pp references cannot cross a Section boundary. Any forward or backward reference to a nonexistent note-statement will be given the value zero. E.g.: the statements i1 i1 i1 0 1 1 1 1 1 10 20 30 np4 pp5
will result in i1 i1 i1 0 1 2 1 1 1 10 20 30 20 30 0 0 20 30
np and pp symbols can provide an instrument with contextual knowledge of the score, enabling it to glissando or crescendo, for instance, toward the pitch or dynamic of some future event (which may or may not be immediately adjacent). Note that while the Carry feature will propagate np and pp through unsorted statements, the operation that interprets these symbols is acting on a time-warped and fully sorted version of the score.
Page 69-3
69.3
Ramping
i statement pfields containing the symbol < will be replaced by values derived from linear interpolation of a time-based ramp. Ramps are anchored at each end by the first real number found in the same pfield of a preceding and following note played by the same instrument. E.g.: the statements i1 i1 i1 i1 i1 i1 0 1 2 3 4 5 1 1 1 1 1 1 100 < < 400 < 0
will result in i1 i1 i1 i1 i1 i1 0 1 2 3 4 5 1 1 1 1 1 1 100 200 300 400 200 0
Ramps cannot cross a Section boundary. Ramps cannot be anchored by an np or pp symbol (although they may be referenced by these). Ramp symbols are illegal in p1, p2 and p3. Ramp symbols may be Carried. Note, however, that while the Carry feature will propagate ramp symbols through unsorted statements, the operation that interprets these symbols is acting on a time-warped and fully sorted version of the score. In fact, time-based linear interpolation is based on warped score-time, so that a ramp which spans a group of accelerating notes will remain linear with respect to strict chronological time. Starting with Csound version 3.52, using the symbols ( or ) will result in an exponential interpolation ramp, similar to expon. The symbols { and } to define an exponential ramp have been deprecated. Using the symbol ~ will result in uniform, random distribution between the first and last values of the ramp. Use of these functions must follow the same rules as the linear ramp function.
Page 69-4
69.4
Score Macros
#define #define $NAME. $NAME. #undef NAME # replacement text # NAME(a b c) # replacement text # NAME
DESCRIPTION
Macros are textual replacements which are made in the score as it is being presented to the system. The macro system in Csound is a very simple one, and uses the characters # and $ to define and call macros. This can allow for simpler score writing, and provide an elementary alternative to full score generation systems. The score macro system is similar to, but independent of, the macro system in the orchestra language. #define NAME defines a simple macro. The name of the macro must begin with a letter and can consist of any combination of letters and numbers. Case is significant. This form is limiting, in that the variable names are fixed. More flexibility can be obtained by using a macro with arguments, described below. #define NAME(a b c) defines a macro with arguments. This can be used in more complex situations. The name of the macro must begin with a letter and can consist of any combination of letters and numbers. Within the replacement text, the arguments can be substituted by the form: $A. In fact, the implementation defines the arguments as simple macros. There may be up to 5 arguments, and the names may be any choice of letters. Remember that case is significant in macro names. $NAME. calls a defined macro. To use a macro, the name is used following a $ character. The name is terminated by the first character which is neither a letter nor a number. If it is necessary for the name not to terminate with a space, a period, which will be ignored, can be used to terminate the name. The string, $NAME., is replaced by the replacement text from the definition. The replacement text can also include macro calls. #undef NAME undefines a macro name. If a macro is no longer required, it can be undefined with #undef NAME.
# replacement text # The replacement text is any character string (not containing a #) and can extend over multiple lines. The replacement text is enclosed within the # characters, which ensure that additional characters are not inadvertently captured.
PERFORMANCE
Some care is needed with textual replacement macros, as they can sometimes do strange things. They take no notice of any meaning, so spaces are significant. This is why, unlike the C programming language, the definition has the replacement text surrounded by # characters. Used carefully, this simple macro system is a powerful concept, but it can be abused.
ANOTHER USE FOR MACROS

When writing a complex score it is sometimes all too easy to forget to what the various instrument numbers refer. One can use macros to give names to the numbers. For example:
#define Flute #define Whoop $Flute. $Flute $Whoop. $Whoop 0 5 #i1# #i2# 10 1 4000 440
Page 69-5
E X A M PL E S
Simple Macro a note-event has a set of p-fields which are repeated:
#define ARGS # 1.01 2.33 138# i1 0 1 8.00 1000 $ARGS i1 0 1 8.01 1500 $ARGS i1 0 1 8.02 1200 $ARGS i1 0 1 8.03 1000 $ARGS This will get expanded before sorting into: i1 0 1 8.00 1000 1.01 2.33 138 i1 0 1 8.01 1500 1.01 2.33 138 i1 0 1 8.02 1200 1.01 2.33 138 i1 0 1 8.03 1000 1.01 2.33 138
This can save typing, and is makes revisions easier. If there were two sets of p-fields one could have a second macro (there is no real limit on the number of macros one can define).
#define ARGS1 # 1.01 2.33 138# #define ARGS2 # 1.41 10.33 1.00# i1 0 1 8.00 1000 $ARGS1 i1 0 1 8.01 1500 $ARGS2 i1 0 1 8.02 1200 $ARGS1 i1 0 1 8.03 1000 $ARGS2
Macros with arguments

1.03 #define ARG(A) # 2.345 i1 0 1 8.00 1000 $ARG(2.0) i1 + 1 8.01 1200 $ARG(3.0) which expands to i1 0 1 8.00 1000 2.345 1.03 i1 + 1 8.01 1200 2.345 1.03 $A 234.9#
2.0 3.0
234.9 234.9
AUTHOR
Page 69-6
69.5
Multiple File Score

It is sometimes convenient to have the score in more than one file. This use is supported by the #include facility which is part of the macro system. A line containing the text #include filename where the character can be replaced by any suitable character. For most uses the double quote symbol will probably be the most convenient. The file name can include a full path. This takes input from the named file until it ends, when input reverts to the previous input. There is currently a limit of 20 on the depth of included files and macros. A suggested use of #include would be to define a set of macros which are part of the composers style. It could also be used to provide repeated sections. s #include section1 ;; Repeat that s #include section1 Alternative methods of doing repeats, use the r, m, and n statements.
AUTHOR
Page 69-7
69.6
Evaluation of Expressions
In earlier versions of Csound the numbers presented in a score were used as given. There are occasions when some simple evaluation would be easier. This need is increased when there are macros. To assist in this area the syntax of an arithmetic expressions within square brackets [ ] has been introduced. Expressions built from the operations +, -, *, /, %, and ^ are allowed, together with grouping with ( ). The expressions can include numbers, and naturally macros whose values are numeric or arithmetic strings. All calculations are made in floating point numbers. Note that unary minus is not yet supported. New in Csound version 3.56 are @x (next power-of-two greater than or equal to x) and @@x (next power-of-two-plus-one greater than or equal to x).
E X A M PL E
r3 i1 i1 e CNT 0 + [0.3*$CNT.] [($CNT./3)+0.2]
As the three copies of the section have the macro $CNT. with the different values of 1, 2 and 3, this expands to
s i1 i1 s i1 i1 s i1 i1 e 0 0.3 0 0.6 0 0.9 0.3 0.533333 0.6 0.866667 0.9 1.2
This is an extreme form, but the evaluation system can be used to ensure that repeated sections are subtly different.
AUTHOR
Page 69-8
69.7
f Statement (or Function Table Statement)

f p1 p2 p3 p4 ...
DESCRIPTION
This causes a GEN subroutine to place values in a stored function table for use by instruments.
P F I E L DS
p1 p2 p3 p4 p5 p6 | .| .| Table number (from 1 to 200) by which the stored function will be known. A negative number requests that the table be destroyed. Action time of function generation (or destruction) in beats. Size of function table (i.e. number of points) Must be a power of 2, or a power-of-2 plus 1 (see below). Maximum table size is 16777216 (2**24) points. Number of the GEN routine to be called (see GEN ROUTINES). A negative value will cause rescaling to be omitted. | | Parameters whose meaning is determined by the particular GEN routine.
SPECIAL CONSIDERATIONS
Function tables are arrays of floating-point values. Arrays can be of any length in powers of 2; space allocation always provides for 2**n points plus an additional guard point. The guard point value, used during interpolated lookup, can be automatically set to reflect the tables purpose: If size is an exact power of 2, the guard point will be a copy of the first point; this is appropriate for interpolated wrap-around lookup as in oscili, etc., and should even be used for non-interpolating oscil for safe consistency. If size is set to 2**n + 1, the guard point value automatically extends the contour of table values; this is appropriate for single-scan functions such in envplx, oscil1, oscil1i, etc. Table space is allocated in primary memory, along with instrument data space. The maximum table number has a soft limit of 200; this can be extended if required. An existing function table can be removed by an f statement containing a negative p1 and an appropriate action time. A function table can also be removed by the generation of another table with the same p1. Functions are not automatically erased at the end of a score section.
Page 69-9
p2 action time is treated in the same way as in i statements with respect to sorting and modification by t statements. If an f statement and an i statement have the same p2, the sorter gives the f statement precedence so that the function table will be available during note initialization. An f 0 statement (zero p1, positive p2) may be used to create an action time with no associated action. Such time markers are useful for padding out a score section (see s statement)
Page 69-10
69.8
i Statement (Instrument or Note Statement)

i p1 p2 p3 p4 ...
DESCRIPTION
This statement calls for an instrument to be made active at a specific time and for a certain duration. The parameter field values are passed to that instrument prior to its initialization, and remain valid throughout its Performance.
P F I E L DS
p1 Instrument number (from 1 to 200), usually a non-negative integer. An optional fractional part can provide an additional tag for specifying ties between particular notes of consecutive clusters. A negative p1 (including tag) can be used to turn off a particular `held note. Starting time in arbitrary units called beats. Duration time in beats (usually positive). A negative value will initiate a held note (see also ihold). A zero value will invoke an initialization pass without performance (see also instr). | | Parameters whose significance is determined by the instrument. | |
p2 p3
p4 p5 . .
Beats are evaluated as seconds, unless there is a t statement in this score section or a -t flag in the command line. Starting or action times are relative to the beginning of a section ( see s statement), which is assigned time 0. Note statements within a section may be placed in any order. Before being sent to an orchestra, unordered score statements must first be processed by Sorter, which will reorder them by ascending p2 value. Notes with the same p2 value will be ordered by ascending p1; if the same p1, then by ascending p3. Notes may be stacked, i.e., a single instrument can perform any number of notes simultaneously. (The necessary copies of the instruments data space will be allocated dynamically by the orchestra loader.) Each note will normally turn off when its p3 duration has expired, or on receipt of a MIDI noteoff signal. An instrument can modify its own duration either by changing its p3 value during note initialization, or by prolonging itself through the action of a linenr unit.
Page 69-11
An instrument may be turned on and left to perform indefinitely either by giving it a negative p3 or by including an ihold in its i-time code. If a held note is active, an i statement with matching p1 will not cause a new allocation but will take over the data space of the held note. The new pfields (including p3) will now be in effect, and an i-time pass will be executed in which the units can either be newly initialized or allowed to continue as required for a tied note (see tigoto). A held note may be succeeded either by another held note or by a note of finite duration. A held note will continue to perform across section endings (see s statement). It is halted only by turnoff or by an i statement with negative matching p1 or by an e statement. It is possible to have multiple instances (usually, but not necessarily, notes of different pitches) of the same instrument, held simultaneously, via negative p3 values. The instrument can then be fed new parameters from the score. This is useful for avoiding long hard-coded linsegs, and can be accomplished by adding a decimal part to the instrument number. For example, to hold three copies of instrument 10 in a simple chord: i10.1 i10.2 i10.3 0 0 0 -1 -1 -1 7.00 7.04 7.07
Subsequent i statements can refer to the same sounding note instances, and if the instrument definition is done properly, the new p-fields can be used to alter the character of the notes in progress. For example, to bend the previous chord up an octave and release it: i10.1 i10.2 i10.3 1 1 1 1 1 1 8.00 8.04 8.07
The instrument definition has to take this into account, however, especially if clicks are to be avoided (see the example below). Note that the decimal instrument number notation cannot be used in conjunction with real-time MIDI. In this case, the instrument would be monophonic while a note was held. Notes being tied to previous instances of the same instrument, should skip most initialization by means of tigoto, except for the values entered in score. For example, all table reading opcodes in the instrument, should usually be skipped, as they store their phase internally. If this is suddenly changed, there will be audible clicks in the output. Note that many opcodes (such as delay and reverb) are prepared for optional initialization. To use this feature, the tival flag is suitable. Therefore, they need not be hidden by a tigoto jump. Beginning with Csound version 3.53, strings are recognized in p- fields for opcodes that accept them (convolve, adsyn, diskin, etc.). There may be only one string per score line.
E X A M PL E
Here is an instrument which can find out whether it is tied to a previous note (tival returns 1), and whether it is held (negative p3). Attack and release are handled accordingly:
instr 10 icps iportime iamp0 iamp1 iamp2 init init init init init cpspch(p4) abs(p3)/7 p5 p5 p5 ;Get target pitch from score event ; Portamento time dep on note length ; Set default amps
Page 69-12
itie if itie iamp0 nofadein: if p3 iamp2
tival == 1 igoto init <
nofadein 0
; Check if this note is tied, ; if not fade in
0 igoto nofadeout 0
init
; Check if this note is held, ; if not fade out
nofadeout: ; Now do amp from the set values: kamp iamp0, .03, iamp1, abs(p3)-.03, iamp2 linseg ; Skip rest of initialization on tied note: tieskip tigoto kcps kcps kpw ar init port oscil vco icps ; Init pitch for untied note icps, iportime, icps ; Drift towards target pitch .4, rnd(1), 1, rnd(.7) ; A simple triangle-saw oscil kamp, kcps, 3, kpw+.5, 1, 1/icps
; (Used in testing - one may set ipch to cpspch(p4+2) ; and view output spectrum) ; ar oscil kamp, kcps, 1 out tieskip: endin ar ; Skip some initialization on tied note
A simple score using three instances of the above instrument:

f1 0 8192 10 1 i10.1 i10.2 i10.3 i10.1 i10.2 i10.3 i10.1 i10.2 i10.3 e 0 0 0 1 1 1 2 2 2 -1 -1 -1 -1 -1 -1 1 1 1 7.00 7.04 7.07 8.00 8.04 8.07 7.11 8.04 8.07 10000 ; Sine
Additional text (Csound version 4.0) explaining tied notes, edited by Rasmus Ekman from a note by David Kirsh, posted to the Csound mailing list. Example instrument by Rasmus Ekman.
Page 69-13
69.9
a Statement (or Advance Statement)

a p1 p2 p3
DESCRIPTION
This causes score time to be advanced by a specified amount without producing sound samples.
P F I E L DS
p1 p2 p3 p4 p5 p6 . . Carries no meaning. Usually zero. Action time, in beats, at which advance is to begin. Number of beats to advance without producing sound. | | These carry no meaning. |
This statement allows the beat count within a score section to be advanced without generating intervening sound samples. This can be of use when a score section is incomplete (the beginning or middle is missing) and the user does not wish to generate and listen to a lot of silence. p2, action time, and p3, number of beats, are treated as in i statements, with respect to sorting and modification by t statements. An a statement will be temporarily inserted in the score by the Score Extract feature when the extracted segment begins later than the start of a Section. The purpose of this is to preserve the beat count and time count of the original score for the benefit of the peak amplitude messages which are reported on the user console. Whenever an a statement is encountered by a performing orchestra, its presence and effect will be reported on the users console.
Page 69-14
69.10 t S t a t e m e n t ( T e m p o S t a t e m e n t )
t p1 p2 p3 p4 ... (unlimited)
DESCRIPTION
This statement sets the tempo and specifies the accelerations and decelerations for the current section. This is done by converting beats into seconds.
P F I E L DS
p1 Must be zero. p2 Initial tempo on beats per minute. p3, p5, p7,... Times in beats per minute (in non-decreasing order). p4, p6, p8,... Tempi for the referenced beat times.
Time and Tempo-for-that-time are given as ordered couples that define points on a tempo vs. time graph. (The time-axis here is in beats so is not necessarily linear.) The beat-rate of a Section can be thought of as a movement from point to point on that graph: motion between two points of equal height signifies constant tempo, while motion between two points of unequal height will cause an accelerando or ritardando accordingly. The graph can contain discontinuities: two points given equal times but different tempi will cause an immediate tempo change. Motion between different tempos over non-zero time is inverse linear. That is, an accelerando between two tempos M1 and M2 proceeds by linear interpolation of the singlebeat durations from 60/M1 to 60/M2. The first tempo given must be for beat 0. A tempo, once assigned, will remain in effect from that time-point unless influenced by a succeeding tempo, i.e. the last specified tempo will be held to the end of the section. A t statement applies only to the score section in which it appears. Only one t statement is meaningful in a section; it can be placed anywhere within that section. If a score section contains no t statement, then beats are interpreted as seconds (i.e. with an implicit t 0 60 statement). N.B. If the Csound command includes a -t flag, the interpreted tempo of all score t statements will be overridden by the command-line tempo.
Page 69-15
69.11 b S t a t e m e n t
b p1
DESCRIPTION
This statement resets the clock for subsequent i statements.
P F I E L DS
p1 Specifies how the clock is to be set.
p1 is the number of beats by which p2 values of subsequent i statements are modified. If p1 is positive, the clock is reset forward, and subsequent notes appear later, the number of beats specified by p1 being added to the notes p2. If p1 is negative, the clock is reset backward, and subsequent notes appear earlier, the number of beats specified by p1 being subtracted from the notes p2. There is no cumulative affect. The clock is reset with each b statement. If p1 = 0, the clock is returned to its original position, and subsequent notes appear at their specified p2.
E X A M PL E
i1 i1 b 5 i2 i2 0 10 1 2 2 888 1 1 440 480 3.1415 1.1111 ; set the clock "forward" ; start time = 6 ; start time = 7 ; set the clock back ; start time = 2 ; start time = 4.5 ; reset clock to normal ; start time = 10
b 1 i3 3 2 i3 5.5 1 b 0 i4 10
200 7
Explanation suggested and example provided by Paul Winkler. (Csound version 4.0)
Page 69-16
69.12 v S t a t e m e n t
v p1
DESCRIPTION
The v statement provides for locally variable time warping of score events.
P F I E L DS
p1 Time warp factor (must be positive).
The v statement takes effect with the following i statement, and remains in effect until the next v, s, or e statement.
E X A M PL E S
The value of p1 is used as a multiplier for the start times (p2) of subsequent i statements.
i1 v2 i1 0 1 1 1 ;note1 ;note2
In this example, the second note occurs two beats after the first note, and is twice as long. Although the v statement is similar to the t statement, the v statement is local in operation. That is, v affects only the following notes, and its effect may be cancelled or changed by another v statement. Carried values (see Section 14.1.1) are unaffected by the v statement.
i1 v2 i1 i1 v1 i1 i1 e 0 1 1 . 2 . 3 . 4 . ;note1 ;note2 ;note3 ;note4 ;note5
In this example, note2 and note4 occur simultaneously, while note3 actually occurs before note2, that is, at its original place. Durations are unaffected.
i1 v2 i. i. 0 1 + . . .
In this example, the v statement has no effect.
Page 69-17
69.13 s S t a t e m e n t
s anything
DESCRIPTION
The s statement marks the end of a section.
P F I E L DS
All p-fields are ignored.
Sorting of the i, f and a statements by action time is done section by section. Time warping for the t statement is done section by section. All action times within a section are relative to its beginning. A section statement establishes a new relative time of 0, but has no other reinitializing effects (e.g. stored function tables are preserved across section boundaries). A section is considered complete when all action times and finite durations have been satisfied (i.e., the length of a section is determined by the last occurring action or turnoff). A section can be extended by the use of an f0 statement. A section ending automatically invokes a Purge of inactive instrument and data spaces. Note: Since score statements are processed section by section, the amount of memory required depends on the maximum number of score statements in a section. Memory allocation is dynamic, and the user will be informed as extra memory blocks are requested during score processing. For the end of the final section of a score, the s statement is optional; the e statement may be used instead.
Page 69-18
69.14 e S t a t e m e n t
e anything
DESCRIPTION
This statement may be used to mark the end of the last section of the score.
P F I E L DS
All pfields are ignored.
The e statement is contextually identical to an s statement. Additionally, the e statement terminates all signal generation (including indefinite performance) and closes all input and output files. If an e statement occurs before the end of a score, all subsequent score lines will be ignored. The e statement is optional in a score file yet to be sorted. If a score file has no e statement, then Sort processing will supply one.
Page 69-19
69.15 r S t a t e m e n t ( R e p e a t S t a t e m e n t )
r p1 p2
DESCRIPTION
Starts a repeated section, which lasts until the next s, r or e statement.
P F I E L DS
p1 p2 Number of times to repeat the section. Macro(name) to advance with each repetition (optional).
In order that the sections may be more flexible than simple editing, the macro named p2 is given the value of 1 for the first time through the section, 2 for the second, and 3 for the third. This can be used to change p-field parameters, or ignored. WARNING: Because of serious problems of interaction with macro expansion, sections must start and end in the same file, and not in a macro.
E X A M PL E
In the following example, the section is repeated 3 times. The macro NN is used and advanced with each repetition.
r3 NN some code ;start of repeated section use macro NN . . . ;end repeat - go back to previous r if repetitions < 3
AUTHOR
Page 69-20
69.16 m S t a t e m e n t ( M a r k S t a t e m e n t )
m p1
DESCRIPTION
Sets a named mark in the score, which can be referenced by an n statement.
P F I E L DS
p1 Name of mark.
This can be helpful in setting a up verse and chorus structure in the score. Names may contain letters and numerals.
AUTHOR
Page 69-21
69.17 n S t a t e m e n t
n p1
DESCRIPTION
Repeats a section from the referenced m statement.
P F I E L DS
p1 Name of mark to repeat.
This can be helpful in setting a up verse and chorus structure in the score. Names may contain letters and numerals.
AUTHOR
Page 69-22
70
GEN R O U T I N E S
The GEN subroutines are function-drawing procedures called by f statements to construct stored wavetables. They are available throughout orchestra performance, and can be invoked at any point in the score as given by p2. p1 assigns a table number, and p3 the table size ( see f statement). p4 specifies the GEN routine to be called; each GEN routine will assign special meaning to the pfield values that follow.
GEN Routines
Page 70-1
70.1
f #
GEN01
time size 1 filcod skiptime format channel
DESCRIPTION
This subroutine transfers data from a soundfile into a function table.
size number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement), with the one exception: allocation of table size can be deferred by setting this parameter to 0. See Notes, below. The maximum table size is 16777216 (224) points. Reading stops at end-of-file or when the table is full. Table locations not filled will contain zeros. filcod integer or character-string denoting the source soundfile name. An integer denotes the file soundin.filcod ; a character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the file is sought first in the current directory, then in that given by the environment variable SSDIR (if defined) then by SFDIR. See also soundin. skiptime begin reading at skiptime seconds into the file. channel channel number to read in. 0 denotes read all channels. An AIFF source can be mono or stereo. format specifies the audio data-file format: 1 - 8-bit signed character 2 - 8-bit A-law bytes 3 - 8-bit U-law bytes 4 - 16-bit short integers 5 - 32-bit long integers 6 - 32-bit floats
If format = 0 the sample format is taken from the soundfile header, or by default from the Csound -o command flag.
NOTES
If the source soundfile is of type AIFF, allocation of table size can be deferred by setting size to 0. The size allocated is then the number of points (or samples) in the file, which is probably not a power-of-2. In this case, the table generated is usable only by loscil Using the form @N for size, where N = the number of samples in the sound file, will give the lowest power of 2 greater than or equal to N. Using the form @@N, adds one to that number, giving a power-of-2 plus 1 sized table. If p4 is positive, the table will be post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped.
GEN Routines
Page 70-2
E X A M PL E S
f f 1 2 0 0 8192 1 23 0 4 0 -1 trumpet A#5 0 4
The tables are filled from 2 files, soundin.23 and trumpet A#5, expected in SSDIR or SFDIR. The first table is pre-allocated; the second is allocated dynamically, and its rescaling is inhibited. .
GEN Routines
Page 70-3
70.2
f #
GEN02
time size 2 v1 v2 v3 . . .
DESCRIPTION
This subroutine transfers data from immediate pfields into a function table.
size number of points in the table. Must be a power of 2 or a power-of-2 plus 1 ( see f statement). The maximum tablesize is 16777216 (224) points. v1, v2, v3, ... values to be copied directly into the table space. The number of values is limited by the compile-time variable PMAX, which controls the maximum pfields (currently 150). The values copied may include the table guard point; any table locations not filled will contain zeros.
NOTE
If p4 is positive, the table will be post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped.
E X A M PL E
f 1 0 16 -2 0 1 2 3 4 5 6 7 8 9 10 11 0
This calls upon GEN02 to place 12 values plus an explicit wrap-around guard value into a table of size next-highest power of 2. Rescaling is inhibited.
GEN Routines
Page 70-4
70.3
f #
GEN03
time size 3 xval1 xval2 c0 c1 c2 . . . cn
DESCRIPTION
This subroutine generates a stored function table by evaluating a polynomial in x over a fixed interval and with specified coefficients.
size number of points in the table. Must be a power of 2 or a power-of-2 plus 1 ( ). xval1, xval2 left and right values of the x interval over which the polynomial is defined (xval1 < xval2). These will produce the 1st stored value and the (power-of-2 plus l)th stored value respectively in the generated function table. c0, c1, c2, ... cn coefficients of the nth-order polynomial c0 + c1x + c2x2 + . . . + cnxn Coefficients may be positive or negative real numbers; a zero denotes a missing term in the polynomial. The coefficient list begins in p7, providing a current upper limit of 144 terms.
NOTE
The defined segment [fn(xval1), fn(xval2)] is evenly distributed. Thus a 512-point table over the interval [-1,1] will have its origin at location 257 (at the start of the 2nd half). Provided the extended guard point is requested, both fn(-1) and fn(1) will exist in the table. GEN03 is useful in conjunction with table or tablei for audio waveshaping (sound modification by non-linear distortion). Coefficients to produce a particular formant from a sinusoidal lookup index of known amplitude can be determined at preprocessing time using algorithms such as Chebyshev formulae. See also GEN13.
E X A M PL E
f 1 0 1025 3 -1 1 5 4 3 2 2 1
This calls GEN03 to fill a table with a 4th order polynomial function over the x-interval -1 to 1. The origin will be at the offset position 512. The function is post-normalized.
GEN Routines
Page 70-5
70.4
f #
GEN04
time size 4 source# sourcemode
DESCRIPTION
This subroutine generates a normalizing function by examining the contents of an existing table.
size number of points in the table. Should be power-of-2 plus 1. Must not exceed (except by 1) the size of the source table being examined; limited to just half that size if the sourcemode is of type offset (see below). source # table number of stored function to be examined. sourcemode a coded value, specifying how the source table is to be scanned to obtain the normalizing function. Zero indicates that the source is to be scanned from left to right. Non-zero indicates that the source has a bipolar structure; scanning will begin at the midpoint and progress outwards, looking at pairs of points equidistant from the center.
NOTE
The normalizing function derives from the progressive absolute maxima of the source table being scanned. The new table is created left-to-right, with stored values equal to 1/(absolute maximum so far scanned). Stored values will thus begin with 1/(first value scanned), then get progressively smaller as new maxima are encountered. For a source table which is normalized (values <= 1), the derived values will range from 1/(first value scanned) down to 1. If the first value scanned is zero, that inverse will be set to 1. The normalizing function from GEN04 is not itself normalized. GEN04 is useful for scaling a table-derived signal so that it has a consistent peak amplitude. A particular application occurs in waveshaping when the carrier (or indexing) signal is less than full amplitude.
E X A M PL E
f 2 0 512 4 1 1
This creates a normalizing function for use in connection with the GEN03 table 1 example. Midpoint bipolar offset is specified.
GEN Routines
Page 70-6
70.5
f f # #
GEN05, GEN07
time time size size 5 7 a a n1 n1 b b n2 n2 c c . . . . . .
DESCRIPTION
These subroutines are used to construct functions from segments of exponential curves (GEN05) or straight lines (GEN07).
size number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). a, b, c, etc. ordinate values, in odd-numbered pfields p5, p7, p9, . . . For GEN05 these must be nonzero and must be alike in sign. No such restrictions exist for GEN07. n1, n2, etc. length of segment (no. of storage locations), in even-numbered pfields. Cannot be negative, but a zero is meaningful for specifying discontinuous waveforms (e.g. in the example below). The sum n1 + n2 + .... will normally equal size for fully specified functions. If the sum is smaller, the function locations not included will be set to zero; if the sum is greater, only the first size locations will be stored.
NOTE
If p4 is positive, functions are post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped. Discrete-point linear interpolation implies an increase or decrease along a segment by equal differences between adjacent locations; exponential interpolation implies that the progression is by equal ratio. In both forms the interpolation from a to b is such as to assume that the value b will be attained in the n + 1th location. For discontinuous functions, and for the segment encompassing the end location, this value will not actually be reached, although it may eventually appear as a result of final scaling.
E X A M PL E
f 1 0 256 7 0 128 1 0 -1 128 0
This describes a single-cycle sawtooth whose discontinuity is mid-way in the stored function.
GEN Routines
Page 70-7
70.6
f #
GEN06
time size 6 a n1 b n2 c n3 d . . .
DESCRIPTION
This subroutine will generate a function comprised of segments of cubic polynomials, spanning specified points just three at a time.
size number of points in the table. Must be a power off or power-of-2 plus 1 (see f statement). a, c, e, ... local maxima or minima of successive segments, depending on the relation of these points to adjacent inflexions. May be either positive or negative. b, d, f, ... ordinate values of points of inflexion at the ends of successive curved segments. May be positive or negative. n1, n2, n3... number of stored values between specified points. Cannot be negative, but a zero is meaningful for specifying discontinuities. The sum n1 + n2 + ... will normally equal size for fully specified functions. (for details, see GEN05).
NOTE
GEN06 constructs a stored function from segments of cubic polynomial functions. Segments link ordinate values in groups of 3: point of inflexion, maximum/minimum, point of inflexion. The first complete segment encompasses b, c, d and has length n2 + n3, the next encompasses d, e, f and has length n4 + n5, etc. The first segment (a, b with length n1) is partial with only one inflexion; the last segment may be partial too. Although the inflexion points b, d, f ... each figure in two segments (to the left and right), the slope of the two segments remains independent at that common point (i.e. the 1st derivative will likely be discontinuous). When a, c, e... are alternately maximum and minimum, the inflexion joins will be relatively smooth; for successive maxima or successive minima the inflexions will be comb-like.
E X A M PL E
f 1 0 65 6 0 16 .5 16 1 16 0 16 -1
This creates a curve running 0 to 1 to -1, with a minimum, maximum and minimum at these values respectively. Inflexions are at .5 and 0, and are relatively smooth.
GEN Routines
Page 70-8
70.7
f #
GEN08
time size 8 a n1 b n2 c n3 d . . .
DESCRIPTION
This subroutine will generate a piecewise cubic spline curve, the smoothest possible through all specified points.
size number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). a, b, c ... ordinate values of the function. n1, n2, n3 ... length of each segment measured in stored values. May not be zero, but may be fractional. A particular segment may or may not actually store any values; stored values will be generated at integral points from the beginning of the function. The sum n1 + n2 + ... will normally equal size for fully specified functions.
NOTE
GEN08 constructs a stored table from segments of cubic polynomial functions. Each segment runs between two specified points but depends as well on their neighbors on each side. Neighboring segments will agree in both value and slope at their common point. (The common slope is that of a parabola through that point and its two neighbors). The slope at the two ends of the function is constrained to be zero (flat). Hint: to make a discontinuity in slope or value in the function as stored, arrange a series of points in the interval between two stored values; likewise for a non-zero boundary slope.
E X A M PL E S
f 1 0 65 8 0 16 0 16 1 16 0 16 0
This example creates a curve with a smooth hump in the middle, going briefly negative outside the hump then flat at its ends.
f 2 0 65 8 0 16 0 .1 0 15.9 1 15.9 0 .1 0 16 0
This example is similar, but does not go negative.
GEN Routines
Page 70-9
70.8
f f f # # #
GEN09, GEN10, GEN19

time time time size 9 size 10 size 19 phsb pna str1 pna dcob . stra str2 stra . . phsa str3 phsa pnb strb phsb . . . str4 . . . . dcoa pnb strb \\
DESCRIPTION
These subroutines generate composite waveforms made up of weighted sums of simple sinusoids. The specification of each contributing partial requires 3 pfields using GEN09, 1 using GEN10, and 4 using GEN19.
size number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). pna, pnb, etc. partial no. (relative to a fundamental that would occupy size locations per cycle) of sinusoid a, sinusoid b, etc. Must be positive, but need not be a whole number, i.e., non-harmonic partials are permitted. Partials may be in any order. stra, strb, etc. strength of partials pna, pnb, etc. These are relative strengths, since the composite waveform may be rescaled later. Negative values are permitted and imply a 180 degree phase shift. phsa, phsb, etc. initial phase of partials pna, pnb, etc., expressed in degrees. dcoa, dcob, etc. DC offset of partials pna, pnb, etc. This is applied after strength scaling, i.e. a value of 2 will lift a 2-strength sinusoid from range [-2,2] to range [0,4] (before later rescaling). str1, str2, str3, etc. relative strengths of the fixed harmonic partial numbers 1,2,3, etc., beginning in p5. Partials not required should be given a strength of zero.
NOTE
These subroutines generate stored functions as sums of sinusoids of different frequencies. The two major restrictions on GEN10 that the partials be harmonic and in phase do not apply to GEN09 or GEN19. In each case the composite wave, once drawn, is then rescaled to unity if p4 was positive. A negative p4 will cause rescaling to be skipped.
E X A M PL E S
f f 1 2 0 0 1024 1024 9 1 3 0 3 1 0 19 .5 1 270 1 9 .3333 180
f 1 combines partials l, 3 and 9 in the relative strengths in which they are found in a square wave, except that partial 9 is upside down. f 2 creates a rising sigmoid [0 2]. Both will be rescaled.
GEN Routines
Page 70-10
70.9
f #
GEN11
time size 11 nh [lh [r]]
DESCRIPTION
This subroutine generates an additive set of cosine partials, in the manner of Csound generators buzz and gbuzz.
size number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement). nh number of harmonics requested. Must be positive. lh (optional) lowest harmonic partial present. Can be positive, zero or negative. The set of partials can begin at any partial number and proceeds upwards; if lh is negative, all partials below zero will reflect in zero to produce positive partials without phase change (since cosine is an even function), and will add constructively to any positive partials in the set. The default value is 1 r (optional) multiplier in an amplitude coefficient series. This is a power series: if the lhth partial has a strength coefficient of A the (lh + n)th partial will have a coefficient of A * rn, i.e. strength values trace an exponential curve. r may be positive, zero or negative, and is not restricted to integers. The default value is 1.
NOTE
This subroutine is a non-time-varying version of the Csound buzz and gbuzz generators, and is similarly useful as a complex sound source in subtractive synthesis. With lh and r present it parallels gbuzz; with both absent or equal to 1 it reduces to the simpler buzz (i.e. nh equal-strength harmonic partials beginning with the fundamental). Sampling the stored waveform with an oscillator is more efficient than using dynamic buzz units. However, the spectral content is invariant, and care is necessary lest the higher partials exceed the Nyquist during sampling to produce foldover.
E X A M PL E S
f f f 1 2 3 0 0 0 2049 2049 2049 11 11 -11 4 4 7 1 3 1 .5
The first two tables will contain identical band-limited pulse waves of four equal-strength harmonic partials beginning with the fundamental. The third table will sum seven consecutive harmonics, beginning with the third, and at progressively weaker strengths (1, .5, .25, .125 . . .). It will not be post-normalized.
GEN Routines
Page 70-11
70.10 G E N 1 2
f # time size -12 xint
DESCRIPTION
This generates the log of a modified Bessel function of the second kind, order 0, suitable for use in amplitude-modulated FM.
size number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. xint specifies the x interval [0 to +int] over which the function is defined.
NOTE
This subroutine draws the natural log of a modified Bessel function of the second kind, order 0 (commonly written as I subscript 0), over the x-interval requested. The call should have rescaling inhibited. The function is useful as an amplitude scaling factor in cycle-synchronous amplitudemodulated FM. (See Palamin & Palamin, J. Audio Eng. Soc., 36/9, Sept. 1988, pp.671-684.) The algorithm is interesting because it permits the normally symmetric FM spectrum to be made asymmetric around a frequency other than the carrier, and is thereby useful for formant positioning. By using a table lookup index of I(r 1/r), where I is the FM modulation index and r is an exponential parameter affecting partial strengths, the Palamin algorithm becomes relatively efficient, requiring only oscils, table lookups, and a single exp call.
E X A M PL E
f 1 0 2049 -12 20
This draws an unscaled ln(I0(x)) from 0 to 20.
GEN Routines
Page 70-12
70.11 G E N 1 3 , G E N 1 4
f f # # time time size size 13 14 xint xint xamp xamp h0 h0 h1 h1 h2 h2 . . . . . . hn hn
DESCRIPTION
These subroutines use Chebyshev coefficients to generate stored polynomial functions which, under waveshaping, can be used to split a sinusoid into harmonic partials having a pre-definable spectrum.
size number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. xint provides the left and right values [-xint, +xint] of the x interval over which the polynomial is to be drawn. These subroutines both call GEN03 to draw their functions; the p5 value here is therefor expanded to a negative-positive p5,p6 pair before GEN03 is actually called. The normal value is 1. xamp amplitude scaling factor of the sinusoid input that is expected to produce the following spectrum. h0, h1, h2, .... hn relative strength of partials 0 (DC), 1 (fundamental), 2 ... that will result when a sinusoid of amplitude
xamp * int(size/2)/xint
is waveshaped using this function table. These values thus describe a frequency spectrum associated with a particular factor xamp of the input signal.
NOTE:
GEN13 is the function generator normally employed in standard waveshaping. It stores a polynomial whose coefficients derive from the Chebyshev polynomials of the first kind, so that a driving sinusoid of strength xamp will exhibit the specified spectrum at output. Note that the evolution of this spectrum is generally not linear with varying xamp. However, it is bandlimited (the only partials to appear will be those specified at generation time); and the partials will tend to occur and to develop in ascending order (the lower partials dominating at low xamp, and the spectral richness increasing for higher values of xamp). A negative hn value implies a 180 degree phase shift of that partial; the requested fullamplitude spectrum will not be affected by this shift, although the evolution of several of its component partials may be. The pattern +,+,-,-,+,+,... for h0,h1,h2... will minimize the normalization problem for low xamp values (see above), but does not necessarily provide the smoothest pattern of evolution. GEN14 stores a polynomial whose coefficients derive from Chebyshevs of the second kind.
GEN Routines
Page 70-13
E X A M PL E
f 1 0 1025 13 1 1 0 5 0 3 0 1
This creates a function which, under waveshaping, will split a sinusoid into 3 odd-harmonic partials of relative strength 5:3:1.
GEN Routines
Page 70-14
70.12 G E N 1 5
f # time size 15 phs2 . xint . . xamp h0 phs0 h1 phs1 h2
DESCRIPTION
This subroutine creates two tables of stored polynomial functions, suitable for use in phase quadrature operations.
size number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. xint provides the left and right values [-xint, +xint] of the x interval over which the polynomial is to be drawn. This subroutine will eventually call GEN03 to draw both functions; this p5 value is therefor expanded to a negative-positive p5, p6 pair before GEN03 is actually called. The normal value is 1. xamp amplitude scaling factor of the sinusoid input that is expected to produce the following spectrum. h0, h1, h2, ... hn relative strength of partials 0 (DC), 1 (fundamental), 2 ... that will result when a sinusoid of amplitude
xamp * int(size/2)/xint
is waveshaped using this function table. These values thus describe a frequency spectrum associated with a particular factor xamp of the input signal. phs0, phs1, ... phase in degrees of desired harmonics h0, h1, ... when the two functions of GEN15 are used with phase quadrature.
NOTE
GEN15 creates two tables of equal size, labeled f # and f # + 1. Table # will contain a Chebyshev function of the first kind, drawn using GEN03 with partial strengths h0cos(phs0), h1cos(phs1), ... Table #+1 will contain a Chebyshev function of the 2nd kind by calling GEN14 with partials h1sin(phs1), h2sin(phs2),... (note the harmonic displacement). The two tables can be used in conjunction in a waveshaping network that exploits phase quadrature.
GEN Routines
Page 70-15
70.13 G E N 1 6
f # time size 15 beg dur type end
DESCRIPTION
Creates a table from beg value to end value of dur steps.
size number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. beg starting value dur number of segments type if 0, a straight line is produced. If non-zero, then GEN16 creates the following curve, for dur steps:
beg+(end-beg)*(1-exp(i*type/(dur-1)))/(1-exp(type))
end value after dur segments
NOTES
If type > 0, there is a slowly rising, fast decaying (convex) curve, while if type < 0, the curve is fast rising, slowly decaying (concave). See also transeg.
AUTHOR
GEN Routines
Page 70-16
70.14 G E N 1 7
f # time size 17 x1 a x2 b x3 c . . .
DESCRIPTION
This subroutine creates a step function from given x-y pairs.
size number of points in the table. Must be a power of 2 or a power-of-2 plus 1 (see f statement). The normal value is power-of-2 plus 1. x1, x2, x3, etc. x-ordinate values, in ascending order, 0 first. a, b, c, etc. y-values at those x-ordinates, held until the next x-ordinate.
NOTE
This subroutine creates a step function of x-y pairs whose y-values are held to the right. The right-most y-value is then held to the end of the table. The function is useful for mapping one set of data values onto another, such as MIDI note numbers onto sampled sound ftable numbers ( see loscil).
E X A M PL E
f 1 0 128 -17 0 1 12 2 24 3 36 4 48 5 60 6 72 7 84 8
This describes a step function with 8 successively increasing levels, each 12 locations wide except the last which extends its value to the end of the table. Rescaling is inhibited. Indexing into this table with a MIDI note-number would retrieve a different value every octave up to the eighth, above which the value returned would remain the same.
GEN Routines
Page 70-17
70.15 G E N 2 0
f # time size 20 window max [opt]
DESCRIPTION
This subroutine generates functions of different windows. These windows are usually used for spectrum analysis or for grain envelopes.
size number of points in the table. Must be a power of 2 ( + 1). window Type of window to generate. 1 2 3 4 5 6 7 8 9 = = = = = = = = = Hamming Hanning Bartlett (triangle) Blackman (3-term) Blackman-Harris (4-term) Gaussian Kaiser Rectangle Sync
max For negative p4 this will be the absolute value at window peak point. If p4 is positive or p4 is negative and p6 is missing the table will be post-rescaled to a maximum value of 1. opt Optional argument required by the Kaiser window.
E X A M PL E S
f 1 0 1024 20 5
This creates a function which contains a 4 term Blackman Harris window with maximum value of 1.
f 1 0 1024 -20 2 456
This creates a function that contains a Hanning window with a maximum value of 456.
f 1 0 1024 -20 1
This creates a function that contains a Hamming window with a maximum value of 1.
f 1 0 1024 20 7 1 2
This creates a function that contains a Kaiser window with a maximum value of 1. The extra argument specifies how open the window is, for example a value of 0 results in a rectangular window and a value of 10 in a Hamming like window. For diagrams, see Appendix. Section 76.4.
GEN Routines
Page 70-18
AUTHORS
Paris Smaragdis MIT, Cambridge 1995 John ffitch University of Bath/Codemist Ltd. Bath, UK New in Csound version 3.2
GEN Routines
Page 70-19
70.16 G E N 2 1
f # time size 21 type level [arg1 [arg2]]
DESCRIPTION
This generates tables of different random distributions. (See also x-class noise generators.) time and size are the usual GEN function arguments. level defines the amplitude. Note that GEN21 is not self-normalizing as are most other GEN functions. type defines the distribution to be used as follows: 1 = Uniform (positive numbers only) 2 = Linear (positive numbers only) 3 = Triangle (positive and negative numbers) 4 = Exponential (positive numbers only) 5 = Biexponential (positive and negative numbers) 6 = Gaussian (positive and negative numbers) 7 = Cauchy (positive and negative numbers) 8 = Positive Cauchy (positive numbers only) 9 = Beta (positive numbers only) 10 = Weibull (positive numbers only) 11 = Poisson (positive numbers only)
Of all these cases only 9 (Beta) and 10 (Weibull) need extra arguments. Beta needs two arguments and Weibull one.
E X A M PL E S
f1 f1 f1 f1 0 0 0 0 1024 1024 1024 1024 21 21 21 21 1 6 9 1 1 2 10 1 2 ; ; ; ; Uniform (white noise) Gaussian Beta (note that level precedes arguments) Weibull
All of the above additions were designed by the author between May and December 1994, under the supervision of Dr. Richard Boulanger.
AUTHORS
Paris Smaragdis MIT, Cambridge 1995 John ffitch University of Bath/Codemist Ltd. Bath, UK New in Csound version 3.2
GEN Routines
Page 70-20
70.17 G E N 2 3
f # time size -23 filename.txt
DESCRIPTION
This subroutine reads numeric values from an external ASCII file
filename.txt numeric values contained in filename.txt (which indicates the complete pathname of the character file to be read), can be separated by spaces, tabs, newline characters or commas. Also, words that contains non-numeric characters can be used as comments since they are ignored. size number of points in the table. Must be a power of 2 , power of 2 + 1, or zero. If size = 0, table size is determined by the number of numeric values in filename.txt. (New in Csound version 3.57)
NOTE
All characters following ; (comment) are ignored until next line (numbers too).
AUTHOR
Gabriel Maldonado Italy February, 1998 New in Csound version 3.47
GEN Routines
Page 70-21
70.18 G E N 2 5 , G E N 2 7
f f # # time time size size 25 27 x1 x1 y1 y1 x2 x2 y2 y2 x3 x3 . . . . . .
DESCRIPTION
These subroutines are used to construct functions from segments of exponential curves (GEN25) or straight lines (GEN27) in breakpoint fashion.
size number of points in the table. Must be a power of 2 or power-of-2 plus 1 (see f statement. x1, x2, x3, etc. locations in table at which to attain the following y value. Must be in increasing order. If the last value is less than size, then the rest will be set to zero. Should not be negative but can be zero. y1, y2, y3,, etc. Breakpoint values attained at the location specified by the preceding x value. For GEN25 these must be non-zero and must be alike in sign. No such restrictions exist for GEN27.
NOTE
If p4 is positive, functions are post-normalized (rescaled to a maximum absolute value of 1 after generation). A negative p4 will cause rescaling to be skipped.
E X A M PL E
f 1 0 257 27 0 0 100 1 200 -1 256 0
This describes a function which begins at 0, rises to 1 at the 100th table location, falls to 1, by the 200th location, and returns to 0 by the end of the table. The interpolation is linear.
AUTHOR
John ffitch University of Bath/Codemist Ltd. Bath, UK New in Csound version 3.49
GEN Routines
Page 70-22
70.19 G E N 2 8
f # time size 28 ifilcod
DESCRIPTION
This function generator reads a text file which contains sets of three values representing the xy coordinates and a time-tag for when the signal should be placed at that location, allowing the user to define a time-tagged trajectory. The file format is in the form:
time1 time2 time3 X1 X2 X3 Y1 Y2 Y3
The configuration of the XY coordinates in space places the signal in the following way: a1 is -1, 1 a2 is 1, 1 a3 is -1, -1 a4 is 1, -1. This assumes a loudspeaker set up as a1 is left front, a2 is right front, a3 is left back, a4 is right back. Values greater than 1 will result in sounds being attenuated as if in the distance. GEN28 creates values to 10 milliseconds of resolution.
size number of points in the table. Must be 0. GEN28 takes 0 as the size and automatically allocates memory. ifilcod character-string denoting the source soundfile name. A character-string (in double quotes, spaces permitted) gives the filename itself, optionally a full pathname. If not a full path, the named file is sought in the current directory.
EXAMPLE
f1 0 0 28 move
The file move should look like:

0 1 2 2.1 3 5 -1 1 4 -4 10 -40 1 1 4 -4 -10 0
Since GEN28 creates values to 10 milliseconds of resolution, there will be 500 values created by interpolating X1 to X2 to X3 and so on, and Y1 to Y2 to Y3 and so on, over the appropriate number of values that are stored in the function table. The sound will begin in the left front, over 1 second it will move to the right front, over another second it move further into the distance but still in the left front, then in just 1/10th of a second it moves to the left rear, a bit distant. Finally over the last .9 seconds the sound will move to the right rear, moderately distant, and it comes to rest between the two left channels (due west!), quite distant.
GEN Routines
Page 70-23
AUTHOR
GEN Routines
Page 70-24
71
THE CSOUND COMMAND

Csound is a command for passing an orchestra file and score file to Csound to generate a soundfile. The score file can be in one of many different formats, according to user preference. Translation, sorting, and formatting into orchestra-readable numeric text is handled by various preprocessors; all or part of the score is then sent on to the orchestra. Orchestra performance is influenced by command flags, which set the level of displays and console reports, specify 1/0 filenames and sample formats, and declare the nature of realtime sensing and control.
71.1
Order of Precedence
With some recent additions to Csound, there are now three places (and in some cases four) where options for Csound performance may be set. They are processed in the following order: 1. 2. 3. 4. 5. Csounds own defaults .csoundrc file Csound command line <CsOptions> tag in a .csd file Orchestra header (for sr, kr, ksmps, nchnls)
The last assignment of an option will override any earlier ones.
71.2
Generic Flags
These are generic Csound command flags. Various platform implementations may not react the same way to different flags! The format of a command is: csound [-flags] orchname scorename where the arguments are of 2 types: flag arguments (beginning with a -), and name arguments (such as filenames). Certain flag arguments take a following name or numeric argument. The available flags are: -U -C -I -n -i -o -b -B -A -W -J -h unam run utility program unam use Cscore processing of scorefile i-time only orch run no sound onto disk fnam sound input filename fnam sound output filename N sample frames (or -kprds) per software sound I/O buffer N samples per hardware sound I/O buffer create an AIFF format output soundfile create a WAV format output soundfile create an IRCAM format output soundfile no header on output soundfile
The Csound Command
Page 71-1
-c -a -8 -u -s -l -f -r N -k N -v -m N -d -g -G -S -x fnam -t N -L dnam -M dnam -F fnam -P N -R -H/H1 -H2 -H3 -H4 -N -T -D -z -z1 -- fnam -j fnam -Z -K num
8-bit signed_char sound samples alaw sound samples 8-bit unsigned_char sound samples ulaw sound samples short_int sound samples long_int sound samples float sound samples orchestra srate override orchestra krate override verbose orch translation TTY message level. Sum of: 1=note amps, 2=out-of-range msg, 4=warnings suppress all displays suppress graphics, use ASCII displays suppress graphics, use Postscript displays score is in Scot format extract from score.srt using extract file fnam use uninterpreted beats of the score, initially at tempo N read Line-oriented real-time score events from device dnam read MIDI real-time events from device dnam read MIDI file event stream from file fnam MIDI sustain pedal threshold (N = 0-128) continually rewrite header while writing soundfile (WAV/AIFF) generates a rotating line progress report generates a . every time a buffer is written reports the size of the output in seconds. In Windows, writes the information to the window title bar. sounds a bell for every buffer of the output written notify (ring the bell) when score or MIDI track is done terminate the performance when MIDI track is done defer GEN01 soundfile loads until performance time List opcodes in this version List opcodes with arguments in this version Log all text output to file fnam derive console messages from database fnam Switch on dithering of audio conversion from internal floating point to 32, 16 and 8 bit formats. (New in Csound version 4.05) Switch off peak chunks.
71.3
PC Windows Specific flags

-j num -J num -q num -p num -O -e set the number of console text rows (default 25) set the number of console text columns (default 80) WAVE OUT device id number (use only if more than one WAVE device is installed) number of WAVE OUT buffers (default 4; max. 40) suppresses all console text output for better real-time performance allows any sample rate (use only with WAVE cards supporting this feature)
The Csound Command
Page 71-2
-y -E -Q num -Y -*
doesnt wait for keypress on exit allows graphic display for WCSHELL by Riccardo Bianchini enable MIDI OUT. num (optional) = MIDI OUT port device id number suppresses real-time WAVE OUT for better MIDI OUT timing performance yields control to the system until audio output buffer is full
The Csound Command
Page 71-3
71.4
-q -Q -X -V -E -p -e -w -y -Y
Macintosh Specific Flags

set the directory for finding samples set the directory for finding analyses set the directory for saving sound files set screen buffer size set number of graphs saved play on finishing set rescaling factor set recording of MIDI data set rate for progress display set rate for profile display
sampdir analdir snddir num num num num num
71.5
Description
Flags may appear anywhere in the command line, either separately or bundled together. A flag taking a Name or Number will find it in that argument, or in the immediately subsequent one. The following are thus equivalent commands: csound nm3 orchname Sxxfilename scorename csound n m 3 orchname x xfilename S scorename All flags and names are optional. The default values are: csound s otest b1024 B1024 m7 P128 orchname scorename where orchname is a file containing Csound orchestra code, and scorename is a file of score data in standard numeric score format, optionally presorted and time-warped. If scorename is omitted, there are two default options: if real-time input is expected (-L, -M or -F), a dummy score file is substituted consisting of the single statement f 0 3600 (i.e. listen for RT input for one hour) else Csound uses the previously processed score.srt in the current directory.
Csound reports on the various stages of score and orchestra processing as it goes, doing various syntax and error checks along the way. Once the actual performance has begun, any error messages will derive from either the instrument loader or the unit generators themselves. A Csound command may include any rational combination of the following flag arguments, with default values as described:
Csound -U Invoke Utility Preprocessing programs: sndinfo, hetro, lpanal, pvanal, cvanal, and pvlook. Csound -I i-time only. Allocate and initialize all instruments as per the score, but skip all p-time processing (no k-signals or a-signals, and thus no amplitudes and no sound). Provides a fast validity check of the score pfields and orchestra i-variables.
The Csound Command
Page 71-4
Csound -n No sound. Do all processing, but bypass writing of sound to disk. This flag does not change the execution in any other way. Csound -i isfname Input soundfile name. If not a full pathname, the file will be sought first in the current directory, then in that given by the environment variable SSDIR (if defined), then by SFDIR. The name stdin will cause audio to be read from standard input. If RTAUDIO is enabled, the name devaudio will request sound from the host audio input device. Csound -o osfname Output soundfile name. If not a full pathname, the soundfile will be placed in the directory given by the environment variable SFDIR (if defined), else in the current directory. The name stdout will cause audio to be written to standard output. If no name is given, the default name will be test. If RTAUDIO is enabled, the name devaudio will send to the host audio output device. Csound -b Numb Number of audio sample-frames per sound i/o software buffer. Large is efficient, but small will reduce audio I/O delay. The default is 1024. In real-time performance, Csound waits on audio I/O on Numb boundaries. It also processes audio (and polls for other input like MIDI) on orchestra ksmps boundaries. The two can be made synchronous. For convenience, if Numb = -N (is negative) the effective value is ksmps * N (audio synchronous with k-period boundaries). With N small (e.g. 1) polling is then frequent and also locked to fixed DAC sample boundaries. Csound -B Numb Number of audio sample-frames held in the DAC hardware buffer. This is a threshold on which software audio I/O (above) will wait before returning. A small number reduces audio I/O delay; but the value is often hardware limited, and small values will risk data lattes. The default is 1024. Csound -h No header on output soundfile. Dont write a file header, just binary samples. Csound {-c, -a, -u, -s, -l, -f} Audio sample format of the output soundfile. One of: c = 8-bit signed character a = 8-bit a-law u = 8-bit u-law s = short integer l = long integer f = single-precision float (not playable, but can be read by -i, soundin and GEN01)
The Csound Command
Page 71-5
Csound -A Write an AIFF output soundfile. Restricts the above formats to c, s, l, or f (AIFC). Csound W Write a .wav output soundfile. Csound J Write an IRCAM output soundfile. Csound -v Verbose translate and run. Prints details of orch translation and performance, enabling errors to be more clearly located. Csound -m Numb Message level for standard (terminal) output. Takes the sum of 3 print control flags, turned on by the following values: 1 = note amplitude messages 2 = samples out of range message 4 = warning messages. The default value is m7 (all messages on). Csound -d Suppress all displays. Csound -g Recast graphic displays into ASCII characters, suitable for any terminal. Csound -S Interpret scorename as a Scot format file and create a standard score file (named score) from it, then sort and perform that. Csound -x xfile Extract a portion of the sorted score score.srt, according to xfile (see Extract). Csound -t Numb Use the uninterpreted beats of score.srt for this performance, and set the initial tempo at Numb beats per minute. When this flag is set, the tempo of score performance is also controllable from within the orchestra. The flag t0 will prevent Csound from deleting the sorted score file, score.srt, upon exit.
The Csound Command
Page 71-6
Csound -L devname Read Line-oriented real-time score events from device devname. The name stdin will permit score events to be typed at your terminal, or piped from another process. Each lineevent is terminated by a carriage-return. Events are coded just like those in a standard numeric score, except that an event with p2=0 will be performed immediately, and an event with p2=T will be performed T seconds after arrival. Events can arrive at any time, and in any order. The score carry feature is legal here, as are held notes (p3 negative) and string arguments, but ramps and pp or np references are not. Csound -M devname Read MIDI events from device devname. Csound -F mfname Read MIDI events from MIDI file mfname. Csound -P Numb Set MIDI sustain pedal threshold (0 128). The official switch value of 64 is normally too low, and is more realistic above 100. The default value of 128 will block all pedal info. Csound -N Notify (ring the bell) when score or MIDI track is done. Csound -T Terminate the performance when MIDI track is done. Csound j fnam Use database fnam for messages to print to console during performance. (New in version 3.55) Csound -K num Switch off peak chunks. New in Csound version 4.09.
The Csound Command
Page 71-7
PC/W I N DO W S - S PE C I F I C
Csound -q num
F L A GS
WAVE OUT device id number (optional, use only if WAVE OUT devices are more than one) Csound -p num number of WAVE OUT buffers (optional; default=4, maximum=40). -b (buffer length) and -p flags are related each other. Finding the optimum values for -b and -p flags requires some experimentation: more buffer length means more latency delay but also more safety from dropouts and sound interruptions (flag -B is now obsolete, dont use it). You now can drastically reduce buffer length and delay by using -e flag and rounded sr and kr. Note that sometimes a smaller buffer length can handle sound flow better than a larger. Only experiments can lead you toward optimal -b values. -b and -p flags value can now be reduced considerably by using rounded ar and kr values (for example ar=32000 and kr=320; ar=40000 and kr=400 and so on) together with -e flag. This feature has been tested only with a SB16 ASP and with an AWE32 card. Support by other cards is unknown. Reducing -b and -p flag values means reducing latency delay and so a more interactive real-time playing. Csound -j num console virtual text rows number. Csound -J num console virtual text columns number. Csound -O (uppercase letter) suppresses all printf for better real-time performance. This switch is better than -m0 because -m0 still leaves some message output to the console. Use both switches together for maximum performance speed. Csound -e allows arbitrary output sample rate (for cards that support this feature). Csound -y doesnt wait for keypress on exit.
The Csound Command
Page 71-8
Csound -E graphic display for WCSHELL by Riccardo Bianchini. Csound -Q num enables MIDI OUT operations and optionally chooses device id num (if num argument is present). This flag allows parallel MIDI OUT and DAC performance. Unfortunately the realtime timing implemented in Csound is completely managed by DAC buffer sample flow. So MIDI OUT operations can present some time irregularities. These irregularities can be fully eliminated when suppressing DAC operations themselves (see -Y flag). Csound -Y disables WAVE OUT (for better MIDI OUT timing performances). This enhances timing of MIDI out operations when used in conjunction with -Q flag. Low k-rates (max. kr=1000) are recommended for use with the -Y flag. As in Win95 maximum timer resolution is 1/1000 of second, unpredictable results can occur when using it at k-rates greater than 1000. Also it is very important to set only kr values in which the following division: 1000/kr produces integer results (some example: kr = 10; 20; 50; 100; 125; 200; 250 etc.) because Win95 timer only handles integer periods in milliseconds. If you use a kr value that produces a non-integer result in the above formula, Csound seems to run normally but times will be not reliable. A value of kr=200 works well on most computers. Maybe with slower computers a lower value works better. Experiment! Values greater than 200 increase the overhead affecting the entire system, and do not give a notable precision improvement. A time resolution of 1/200 of sec is precise enough for almost all MIDI applications. The sr/kr/ksmps ratio must be respected, or an error message will stop the performance, even if sr value is meaningless when using -Y flag. Csound -* compels Csound to yield control to system until audio output buffer content passes a certain threshold. Below this threshold Csound continues processing, while over this threshold Csound yields control to Windows. This gives a big enhancement in multitasking processes. Enabling this option reduces polyphony a bit when using short buffer space. So the user should increase the number (-p flag) and the length (-b flag) of buffers when setting -* flag. Experiment to find best values. Do not use this flag when time response to gestural actions is critical.
The Csound Command
Page 71-9
The Csound Command
Page 71-10
72
UNIFIED FILE FORMAT FOR ORCHESTRAS AND SCORES

Description
The Unified File Format, introduced in Csound version 3.50, enables the orchestra and score files, as well as command line flags, to be combined in one file. The file has the extension .csd. This format was originally introduced by Michael Gogins in AXCsound. The file is a structured data file which uses markup language, similar to any SGML such as HTML. Start tags (<tag>) and end tags (</tag>) are used to delimit the various elements. The file is saved as a text file.
72.1
72.2
Structured Data File Format

The Csound Element is used to alert the csound compiler to the .csd format. The file must begin with the start tag <CsoundSynthesizer>. The last line of the file must be the end tag </CsoundSynthesizer>. The remaining elements are defined below.
M A N DA T O R Y E L E M E N T S
Options Csound command line flags are put in the Options Element. This section is delimited by the start tag <CsOptions> and the end tag </CsOptions> Lines beginning with # or ; are treated as comments. For precedence of flags, options, and header statements, see Section 67.1. Instruments (Orchestra) The instrument definitions (orchestra) are put into the Instruments Element. The statements and syntax in this section are identical to the Csound orchestra file, and have the same requirements, including the header statements (sr, kr, etc.) This Instruments Element is delimited with the start tag <CsInstruments> and the end tag </CsInstruments>. Score
Csound score statements are put in the Score Element. The statements and syntax in this section are identical to the Csound score file, and have the same requirements. The Score Element is delimited by the start tag <CsScore> and the end tag </CsScore>.
O PT I O N A L E L E M E N T S
Included Base64 Files Base64 encoded MIDI files may be included with the tag <CsMidifileB filename=filename>, where filename is the name of the file containing the MIDI information. There is no matching end tag. New in Csound version 4.07.
Unified File Format for Orchestras and Scores
Page 72-1
Base64 encoded sample files may be included with the tag <CsSampleB filename=filename>, where filename is the name of the file containing the sample. There is no matching end tag. New in Csound version 4.07. Version Blocking Versions of Csound may blocked by placing one of the following statements between the start tag <CsVersion> and the end tag </CsVersion>: Before #.# or After #.# where #.# is the requested Csound version number. The second statement may be written simply as: #.# See example below. New in Csound version 4.09.
72.3
Example
Below is a sample file, test.csd, which renders a .wav file at 44.1 kHz sample rate containing one second of a 1 kHz sine wave. Displays are suppressed. test.csd was created from two files, tone.orc and tone.sco, with the addition of command line flags.
<CsoundSynthesizer> ; test.csd a Csound structured data file <CsOptions> -W -d -o tone.wav </CsOptions> ;optional section <CsVersion> Before 4.10 ;these two statements check for After 4.08 ; Csound version 4.09 </CsVersion> <CsInstruments> ; originally tone.orc sr = 44100 kr = 4410 ksmps = 10 nchnls = 1 instr 1 a1 oscil out endin </CsInstruments>
p4, p5, 1 ; simple oscillator a1
<CsScore> ; originally tone.sco f1 0 8192 10 1 i1 0 1 20000 1000 ;play one second of one kHz tone e </CsScore> </CsoundSynthesizer>
Page 72-2
72.4
Command Line Parameter File

If the file .csoundrc exists, it will be used to set the command line parameters. These can be overridden. It uses the same form as a .csd file. Lines beginning with # or ; are treated as comments.
Page 72-3
Page 72-4
73
73.1
SCORE FILE PREPROCESSING

The Extract Feature
This feature will extract a segment of a sorted numeric score file according to instructions taken from a control file. The control file contains an instrument list and two time points, from and to, in the form: instruments 1 2 from 1:27.5 to 2:2
The component labels may be abbreviated as i, f and t. The time points denote the beginning and end of the extract in terms of: [section no.] : [beat no.]. each of the three parts is also optional. The default values for missing i, f or t are: all instruments, beginning of score, end of score.
Score File Preprocessing
Page 73.1
73.2
Independent Pre-Processing with Scsort

Although the result of all score preprocessing is retained in the file score.srt after orchestra performance (it exists as soon as score preprocessing has completed), the user may sometimes want to run these phases independently. The command scot filename will process the Scot formatted filename, and leave a standard numeric score result in a file named score for perusal or later processing. The command scscort < infile > outfile will put a numeric score infile through Carry, Tempo, and Sort preprocessing, leaving the result in outfile. Likewise extract, also normally invoked as part of the Csound command, can be invoked as a standalone program: extract xfile < score.sort > score.extract This command expects an already sorted score. An unsorted score should first be sent through Scsort then piped to the extract program: scsort < scorefile | extract xfile > score.extract
Score File Preprocessing
Page 73.2
74
UTILITY PROGRAMS
The Csound Utilities are soundfile preprocessing programs that return information on a soundfile or create some analyzed version of it for use by certain Csound generators. Though different in goals, they share a common soundfile access mechanism and are describable as a set. The Soundfile Utility programs can be invoked in two equivalent forms: csound U utilname [flags] filenames ... utilname [flags] filenames ... In the first, the utility is invoked as part of the Csound executable, while in the second it is called as a standalone program. The second is smaller by about 200K, but the two forms are identical in function. The first is convenient in not requiring the maintenance and use of several independent programs one program does all. When using this form, a -U flag detected in the command line will cause all subsequent flags and names to be interpreted as per the named utility; i.e. Csound generation will not occur, and the program will terminate at the end of utility processing. Directories. Filenames are of two kinds, source soundfiles and resultant analysis files. Each has a hierarchical naming convention, influenced by the directory from which the Utility is invoked. Source soundfiles with a full pathname (begins with dot (.), slash (/), or for ThinkC includes a colon (:)), will be sought only in the directory named. Soundfiles without a path will be sought first in the current directory, then in the directory named by the SSDIR environment variable (if defined), then in the directory named by SFDIR. An unsuccessful search will return a cannot open error. Resultant analysis files are written into the current directory, or to the named directory if a path is included. It is tidy to keep analysis files separate from sound files, usually in a separate directory known to the SADIR variable. Analysis is conveniently run from within the SADIR directory. When an analysis file is later invoked by a Csound generator it is sought first in the current directory, then in the directory defined by SADIR. Soundfile Formats. Csound can read and write audio files in a variety of formats. Write formats are described by Csound command flags. On reading, the format is determined from the soundfile header, and the data automatically converted to floating-point during internal processing. When Csound is installed on a host with local soundfile conventions (SUN, NeXT, Macintosh) it may conditionally include local packaging code which creates soundfiles not portable to other hosts. However, Csound on any host can always generate and read AIFF files, which is thus a portable format. Sampled sound libraries are typically AIFF, and the variable SSDIR usually points to a directory of such sounds. If defined, the SSDIR directory is in the search path during soundfile access. Note that some AIFF sampled sounds have an audio looping feature for sustained performance; the analysis programs will traverse any loop segment once only. For soundfiles without headers, an SR value may be supplied by a command flag (or its default). If both header and flag are present, the flag value will over-ride.
Utility Programs
Page 74-1
When sound is accessed by the audio Analysis programs , only a single channel is read. For stereo or quad files, the default is channel one; alternate channels may be obtained on request.
Author
Dan Ellis MIT Media Lab Cambrige, Massachussetts 1990
Utility Programs
Page 74-2
74.1
sndinfo
DESCRIPTION
get basic information about one or more soundfiles. csound U sndinfo soundfilenames ... or sndinfo soundfilenames ... sndinfo will attempt to find each named file, open it for reading, read in the soundfile header, then print a report on the basic information it finds. The order of search across soundfile directories is as described above. If the file is of type AIFF, some further details are listed first.
E X A M PL E
csound U sndinfo test Bosendorfer/BOSEN mf A0 st foo foo2 where the environment variables SFDIR = /u/bv/sound, and SSDIR = /so/bv/Samples, might produce the following: util SNDINFO: /u/bv/sound/test: srate 22050, monaural, 16 bit shorts, 1.10 seconds headersiz 1024, datasiz 48500 (24250 sample frames) /so/bv/Samples/Bosendorfer/BOSEN mf A0 st: AIFF, 197586 stereo samples, base Frq 261.6 (MIDI 60), sustnLp: mode 1, 121642 to 197454, relesLp: mode 0 AIFF soundfile, looping with modes 1, 0 srate 44100, stereo, 16 bit shorts, 4.48 seconds headersiz 402, datasiz 790344 (197586 sample frames)
/u/bv/sound/foo: no recognizable soundfile header /u/bv/sound/foo2: couldnt find
Utility Programs
Page 74-3
74.2
hetro
DESCRIPTION
hetrodyne filter analysis for the Csound adsyn generator. csound U hetro [flags] infilename outfilename or hetro [flags] infilename outfilename hetro takes an input soundfile, decomposes it into component sinusoids, and outputs a description of the components in the form of breakpoint amplitude and frequency tracks. Analysis is conditioned by the control flags below. A space is optional between flag and value. -s srate sampling rate of the audio input file. This will over-ride the srate of the soundfile header, which otherwise applies. If neither is present, the default is 10000. Note that for adsyn synthesis the srate of the source file and the generating orchestra need not be the same. -c channel channel number sought. The default is 1. -b begin beginning time (in seconds) of the audio segment to be analyzed. The default is 0.0 -d duration duration (in seconds) of the audio segment to be analyzed. The default of 0.0 means to the end of the file. Maximum length is 32.766 seconds. -f begfreq estimated starting frequency of the fundamental, necessary to initialize the filter analysis. The default is 100 (cps). -h partials number of harmonic partials sought in the audio file. Default is 10, maximum is a function of memory available. -M maxamp maximum amplitude summed across all concurrent tracks. The default is 32767. -m minamp amplitude threshold below which a single pair of amplitude/frequency tracks is considered dormant and will not contribute to output summation. Typical values: 128 (48 dB down from full scale), 64 (54 dB down), 32 (60 dB down), 0 (no thresholding). The default threshold is 64 (54 dB down). -n brkpts initial number of analysis breakpoints in each amplitude and frequency track, prior to thresholding (-m) and linear breakpoint consolidation. The initial points are spread evenly over the duration. The default is 256. -l cutfreq substitute a 3rd order Butterworth low-pass filter with cutoff frequency cutfreq (in Hz), in place of the default averaging comb filter. The default is 0 (dont use).
Utility Programs
Page 74-4
E X A M PL E
hetro -s44100 -b.5 -d2.5 -h16 -M24000 audiofile.test adsynfile7
This will analyze 2.5 seconds of channel 1 of a file audiofile.test, recorded at 44.1 kHz, beginning .5 seconds from the start, and place the result in a file adsynfile7. We request just the first 16 harmonics of the sound, with 256 initial breakpoint values per amplitude or frequency track, and a peak summation amplitude of 24000. The fundamental is estimated to begin at 100 Hz. Amplitude thresholding is at 54 dB down. The Butterworth LPF is not enabled.
FILE FORMAT
The output file contains time-sequenced amplitude and frequency values for each partial of an additive complex audio source. The information is in the form of breakpoints (time, value, time, value, ....) using 16-bit integers in the range 0 32767. Time is given in milliseconds, and frequency in Hertz (Hz). The breakpoint data is exclusively non-negative, and the values -1 and -2 uniquely signify the start of new amplitude and frequency tracks. A track is terminated by the value 32767. Before being written out, each track is datareduced by amplitude thresholding and linear breakpoint consolidation. A component partial is defined by two breakpoint sets: an amplitude set, and a frequency set. Within a composite file these sets may appear in any order (amplitude, frequency, amplitude ....; or amplitude, amplitude..., then frequency, frequency,...). During adsyn resynthesis the sets are automatically paired (amplitude, frequency) from the order in which they were found. There should be an equal number of each. A legal adsyn control file could have following format:
-1 -2 -1 -2 -2 -2 -1 -1 time1 time1 time1 time1 time1 time1 time1 time2 value1 value1 value1 value1 value1 value1 value1 value1 ... timeK valueK ... timeL valueL ... timeM valueM ... timeN valueN .......... .......... .......... .......... 32767 32767 32767 32767 ; ; ; ; amplitude frequency amplitude frequency breakpoints breakpoints breakpoints breakpoints for for for for partial partial partial partial 1 1 2 2
; pairable tracks for partials 3 and 4
If the filename passed to hetro has the extension .sdif, data will be written in SDIF format as 1TRC frames of additive synthesis data. The utility program sdif2ads can be used to convert any SDIF file containing a stream of 1TRC data to the Csound adsyn format. New in Csound version 4.08.
Utility Programs
Page 74-5
74.3
lpanal
DESCRIPTION
linear predictive analysis for the Csound lp generators csound U lpanal [flags] infilename outfilename or lpanal [flags] infilename outfilename lpanal performs both lpc and pitch-tracking analysis on a soundfile to produce a timeordered sequence of frames of control information suitable for Csound resynthesis. Analysis is conditioned by the control flags below. A space is optional between the flag and its value. -a [alternate storage] asks lpanal to write a file with filter poles values rather than the usual filter coefficient files. When lpread / lpreson are used with pole files, automatic stabilization is performed and the filter should not get wild. (This is the default in the Windows GUI) Changed by Marc Resibois. -s srate sampling rate of the audio input file. This will over-ride the srate of the soundfile header, which otherwise applies. If neither is present, the default is 10000. -c channel channel number sought. The default is 1. -b begin beginning time (in seconds) of the audio segment to be analyzed. The default is 0.0 -d duration duration (in seconds) of the audio segment to be analyzed. The default of 0.0 means to the end of the file. -p npoles number of poles for analysis. The default is 34, the maximum 50. -h hopsize hop size (in samples) between frames of analysis. This determines the number of frames per second (srate / hopsize) in the output control file. The analysis framesize is hopsize * 2 samples. The default is 200, the maximum 500. -C string text for the comments field of the lpfile header. The default is the null string. -P mincps lowest frequency (in Hz) of pitch tracking. -P0 means no pitch tracking. -Q maxcps highest frequency (in Hz) of pitch tracking. The narrower the pitch range, the more accurate the pitch estimate. The defaults are -P70, -Q200. -v verbosity level of terminal information during analysis. 0 = none 1 = verbose 2 = debug The default is 0.
Utility Programs
Page 74-6
E X A M PL E
lpanal a p26 d2.5 P100 Q400 audiofile.test lpfil22
will analyze the first 2.5 seconds of file audiofile.test, producing srate/200 frames per second, each containing 26-pole filter coefficients and a pitch estimate between 100 and 400 Hertz. Stabilized (-a) output will be placed in lpfil22 in the current directory.
FILE FORMAT
Output is a file comprised of an identifiable header plus a set of frames of floating point analysis data. Each frame contains four values of pitch and gain information, followed by npoles filter coefficients. The file is readable by Csounds lpread. lpanal is an extensive modification of Paul Lanksys lpc analysis programs.
Utility Programs
Page 74-7
74.4
pvanal
DESCRIPTION
Fourier analysis for the Csound pvoc generator csound U pvanal [flags] infilename outfilename or pvanal [flags] infilename outfilename pvanal converts a soundfile into a series of short-time Fourier transform (STFT) frames at regular timepoints (a frequency-domain representation). The output file can be used by pvoc to generate audio fragments based on the original sample, with timescales and pitches arbitrarily and dynamically modified. Analysis is conditioned by the flags below. A space is optional between the flag and its argument. -s srate sampling rate of the audio input file. This will over-ride the srate of the soundfile header, which otherwise applies. If neither is present, the default is 10000. -c channel channel number sought. The default is 1. -b begin beginning time (in seconds) of the audio segment to be analyzed. The default is 0.0 -d duration duration (in seconds) of the audio segment to be analyzed. The default of 0.0 means to the end of the file. -n frmsiz STFT frame size, the number of samples in each Fourier analysis frame. Must be a power of two, in the range 16 to 16384. For clean results, a frame must be larger than the longest pitch period of the sample. However, very long frames result in temporal smearing or reverberation. The bandwidth of each STFT bin is determined by sampling rate / frame size. The default framesize is the smallest power of two that corresponds to more than 20 milliseconds of the source (e.g. 256 points at 10 kHz sampling, giving a 25.6 ms frame). -w windfact Window overlap factor. This controls the number of Fourier transform frames per second. Csounds pvoc will interpolate between frames, but too few frames will generate audible distortion; too many frames will result in a huge analysis file. A good compromise for windfact is 4, meaning that each input point occurs in 4 output windows, or conversely that the offset between successive STFT frames is framesize/4. The default value is 4. Do not use this flag with -h. -h hopsize STFT frame offset. Converse of above, specifying the increment in samples between successive frames of analysis (see also lpanal). Do not use with -w.
E X A M PL E
pvanal asound pvfile
will analyze the soundfile asound using the default frmsiz and windfact to produce the file pvfile suitable for use with pvoc.
Utility Programs
Page 74-8
FILES
The output file has a special pvoc header containing details of the source audio file, the analysis frame rate and overlap. Frames of analysis data are stored as float, with the magnitude and frequency (in Hz) for the first N/2 + 1 Fourier bins of each frame in turn. Frequency encodes the phase increment in such a way that for strong harmonics it gives a good indication of the true frequency. For low amplitude or rapidly moving harmonics it is less meaningful.
DIAGNOSTICS
Prints total number of frames, and frames completed on every 20th.
AUTHOR
Dan Ellis MIT Media Lab Cambrige, Massachussetts 1990
Utility Programs
Page 74-9
74.5
cvanal
DESCRIPTION
Impulse Response Fourier Analysis for convolve operator Csound -U cvanal [flags] infilename outfilename cvanal converts a soundfile into a single Fourier transform frame. The output file can be used by the convolve operator to perform Fast Convolution between an input signal and the original impulse response. Analysis is conditioned by the flags below. A space is optional between the flag and its argument. -s rate sampling rate of the audio input file. This will over-ride the srate of the soundfile header, which otherwise applies. If neither is present, the default is 10000. -c channel channel number sought. If omitted, the default is to process all channels. If a value is given, only the selected channel will be processed. -b begin beginning time (in seconds) of the audio segment to be analyzed. The default is 0.0 -d duration duration (in seconds) of the audio segment to be analyzed. The default of 0.0 means to the end of the file.
E X A M PL E
cvanal asound cvfile
will analyze the soundfile asound to produce the file cvfile for the use with convolve. To use data that is not already contained in a soundfile, a soundfile converter that accepts text files may be used to create a standard audio file, e.g., the .DAT format for SOX. This is useful for implementing FIR filters.
FILES
The output file has a special convolve header, containing details of the source audio file. The analysis data is stored as float, in rectangular (real/imaginary) form. Note: The analysis file is not system independent! Ensure that the original impulse recording/data is retained. If/when required, the analysis file can be recreated.
AUTHOR
Greg Sullivan (Based on algorithm given in Elements Of Computer Music, by F. Richard Moore.
Utility Programs
Page 74-10
74.6
pvlook
DESCRIPTION
View formatted text output of STFT analysis files created with pvanal. csound U pvlook [flags] infilename or pvlook [flags] infilename pvlook reads a file, and frequency and amplitude trajectories for each of the analysis bins, in readable text form. The file is assumed to be an STFT analysis file created by pvanal. By default, the entire file is processed. -bb N begin at analysis bin number N, numbered from 1. Default is 1. -eb N end at analysis bin number N. Defaults to the highest. -bf N begin at analysis frame number N, numbered from 1. Defaults to 1. -ef N end at analysis frame number N. Defaults to the highest. -i prints values as integers. Defaults to floating point.
E X A M PL E
enakis 259% ../csound -U pvlook test.pv Using csound.txt Csound Version 3.57 (Aug 3 1999) util PVLOOK: ; Bins in Analysis: 513 ; First Bin Shown: 1 ; Number of Bins Shown: 513 ; Frames in Analysis: 1184 ; First Frame Shown: 1 ; Number of Data Frames Shown: 1184 Bin 1 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 Freqs.0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 87.891 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000
Utility Programs
Page 74-11
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 -87.891 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 87.891 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000
Utility Programs
Page 74-12
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 Bin 1 0.252 0.248 0.245 0.251 0.249 0.246 0.250 0.251 0.246 0.249 0.252 0.245 0.248 0.252 0.246 0.248 0.251 0.248 0.247 0.252 0.248 0.245 0.252 0.249 0.245 0.251 0.249 0.245 0.249 0.250 0.245 0.248 0.251 0.245 0.248 0.251 0.245 0.247 0.251 0.246 0.247 0.250 0.248 0.245 0.251 0.248 0.245 0.251 0.248 0.245 0.250 0.249 0.178 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 Amps. 0.251 0.247 0.246 0.252 0.248 0.245 0.253 0.248 0.246 0.252 0.249 0.246 0.250 0.250 0.245 0.249 0.251 0.244 0.249 0.251 0.245 0.249 0.251 0.247 0.247 0.252 0.247 0.245 0.252 0.247 0.245 0.251 0.248 0.245 0.250 0.249 0.245 0.249 0.250 0.244 0.249 0.251 0.245 0.248 0.250 0.246 0.247 0.250 0.247 0.246 0.251 0.247 0.008 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.180 0.250 0.244 0.250 0.250 0.245 0.249 0.251 0.247 0.247 0.252 0.247 0.247 0.253 0.247 0.247 0.252 0.248 0.246 0.250 0.250 0.245 0.250 0.250 0.244 0.250 0.250 0.245 0.249 0.251 0.246 0.247 0.251 0.246 0.246 0.252 0.246 0.246 0.251 0.247 0.246 0.250 0.248 0.245 0.249 0.250 0.244 0.250 0.250 0.245 0.249 0.250 0.246 0.000 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.066 0.248 0.246 0.251 0.249 0.245 0.252 0.249 0.245 0.251 0.249 0.246 0.249 0.251 0.246 0.248 0.252 0.246 0.248 0.252 0.246 0.248 0.251 0.247 0.247 0.251 0.249 0.245 0.251 0.249 0.245 0.251 0.249 0.245 0.249 0.250 0.246 0.248 0.251 0.245 0.247 0.251 0.245 0.247 0.250 0.246 0.246 0.250 0.248 0.246 0.251 0.249 0.245 0.000 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.252 0.244 0.249 0.252 0.246 0.249 0.252 0.247 0.247 0.252 0.248 0.246 0.253 0.248 0.246 0.251 0.248 0.246 0.250 0.250 0.245 0.249 0.251 0.244 0.249 0.251 0.245 0.248 0.251 0.247 0.247 0.251 0.248 0.245 0.251 0.247 0.245 0.251 0.248 0.245 0.250 0.249 0.245 0.249 0.250 0.245 0.248 0.251 0.245 0.248 0.250 0.246 0.247 0.000 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.248 0.245 0.250 0.250 0.245 0.251 0.251 0.244 0.250 0.251 0.246 0.249 0.251 0.247 0.248 0.252 0.247 0.247 0.253 0.247 0.247 0.251 0.248 0.246 0.250 0.250 0.245 0.250 0.250 0.244 0.250 0.250 0.245 0.249 0.250 0.246 0.247 0.251 0.246 0.246 0.251 0.247 0.246 0.251 0.247 0.246 0.250 0.249 0.245 0.250 0.249 0.244 0.249 0.000 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.245 0.248 0.253 0.247 0.248 0.252 0.249 0.247 0.252 0.249 0.245 0.252 0.249 0.244 0.251 0.250 0.245 0.250 0.251 0.246 0.248 0.252 0.246 0.247 0.252 0.246 0.247 0.251 0.247 0.247 0.251 0.249 0.245 0.251 0.249 0.245 0.250 0.249 0.245 0.249 0.250 0.246 0.248 0.251 0.246 0.247 0.251 0.245 0.247 0.251 0.246 0.246 0.250 0.000 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.246 0.250 0.251 0.245 0.249 0.252 0.245 0.249 0.252 0.246 0.249 0.252 0.247 0.246 0.252 0.248 0.245 0.252 0.248 0.245 0.252 0.248 0.245 0.249 0.250 0.245 0.248 0.251 0.245 0.248 0.252 0.246 0.248 0.251 0.247 0.247 0.250 0.248 0.245 0.251 0.248 0.245 0.251 0.248 0.246 0.250 0.249 0.245 0.248 0.250 0.245 0.248 0.251 0.000 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.246 0.254 0.247 0.246 0.252 0.249 0.246 0.250 0.251 0.245 0.250 0.251 0.245 0.250 0.251 0.246 0.249 0.251 0.247 0.247 0.252 0.247 0.246 0.252 0.247 0.246 0.251 0.247 0.246 0.250 0.249 0.245 0.249 0.251 0.244 0.249 0.250 0.245 0.248 0.250 0.246 0.247 0.251 0.247 0.246 0.251 0.247 0.246 0.251 0.247 0.246 0.250 0.287 0.000 0.000
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.249 0.251 0.246 0.247 0.253 0.246 0.248 0.253 0.247 0.248 0.252 0.248 0.246 0.251 0.250 0.245 0.251 0.250 0.245 0.251 0.249 0.245 0.249 0.251 0.246 0.248 0.252 0.246 0.246 0.252 0.246 0.247 0.251 0.247 0.246 0.250 0.249 0.245 0.250 0.249 0.245 0.250 0.249 0.245 0.249 0.249 0.246 0.248 0.250 0.246 0.247 0.251 0.331 0.000 0.000
Utility Programs
Page 74-13
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.140 3.292 3.296 3.292 3.294 3.292 3.295 3.292 3.291 3.296 3.288 3.294 3.294 3.290 3.294 3.292 3.293 3.292 3.293 3.293 3.289 3.295 3.291 3.290 3.294 3.290 3.292 3.291 3.292 3.291 3.290 3.294 3.288 3.291 3.293 3.288 3.293 3.291 3.290 3.291 3.290 3.292 3.288 3.293 3.290 3.288 3.293 3.288 3.290 3.291 3.289 3.290 3.289
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 1.265 3.291 3.297 3.293 3.291 3.297 3.290 3.293 3.294 3.291 3.293 3.291 3.295 3.290 3.292 3.294 3.288 3.295 3.292 3.290 3.293 3.291 3.292 3.290 3.293 3.292 3.289 3.295 3.290 3.291 3.294 3.290 3.291 3.291 3.292 3.290 3.290 3.293 3.288 3.292 3.292 3.288 3.293 3.290 3.290 3.290 3.290 3.290 3.288 3.293 3.289 3.289 3.292
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 2.766 3.297 3.292 3.294 3.293 3.294 3.294 3.290 3.296 3.291 3.292 3.296 3.289 3.293 3.293 3.291 3.292 3.291 3.294 3.289 3.293 3.293 3.289 3.295 3.291 3.291 3.293 3.291 3.292 3.290 3.293 3.290 3.290 3.294 3.288 3.291 3.292 3.289 3.292 3.291 3.292 3.289 3.291 3.292 3.287 3.293 3.290 3.288 3.292 3.289 3.290 3.289 3.291
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 3.289 3.295 3.295 3.290 3.297 3.292 3.292 3.296 3.291 3.293 3.292 3.293 3.292 3.291 3.295 3.289 3.293 3.294 3.289 3.294 3.292 3.292 3.291 3.292 3.293 3.289 3.294 3.291 3.289 3.295 3.290 3.291 3.292 3.291 3.291 3.290 3.293 3.289 3.291 3.293 3.288 3.292 3.291 3.289 3.291 3.291 3.291 3.288 3.292 3.291 3.287 3.292 3.289
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 3.296 3.294 3.292 3.295 3.292 3.295 3.292 3.296 3.293 3.290 3.297 3.291 3.292 3.295 3.291 3.293 3.292 3.293 3.291 3.291 3.295 3.288 3.294 3.293 3.289 3.294 3.291 3.292 3.291 3.292 3.292 3.289 3.294 3.290 3.290 3.294 3.289 3.291 3.291 3.291 3.291 3.290 3.292 3.288 3.291 3.292 3.287 3.292 3.290 3.290 3.290 3.290 3.291
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 3.293 3.296 3.290 3.295 3.295 3.290 3.297 3.292 3.293 3.294 3.292 3.294 3.291 3.296 3.292 3.291 3.296 3.289 3.293 3.293 3.291 3.292 3.291 3.294 3.290 3.293 3.293 3.288 3.294 3.291 3.290 3.293 3.290 3.291 3.290 3.293 3.291 3.289 3.294 3.288 3.291 3.292 3.288 3.291 3.290 3.291 3.289 3.290 3.292 3.287 3.292 3.290 3.288
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 3.296 3.291 3.295 3.292 3.294 3.292 3.293 3.295 3.290 3.296 3.293 3.291 3.296 3.291 3.293 3.293 3.293 3.292 3.290 3.295 3.289 3.291 3.294 3.289 3.293 3.292 3.291 3.292 3.291 3.293 3.289 3.293 3.292 3.288 3.294 3.290 3.290 3.292 3.290 3.291 3.290 3.293 3.289 3.290 3.293 3.288 3.291 3.291 3.289 3.291 3.290 3.291
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 3.296 3.292 3.293 3.296 3.288 3.295 3.293 3.291 3.295 3.292 3.294 3.292 3.294 3.294 3.290 3.296 3.291 3.291 3.295 3.290 3.292 3.291 3.292 3.291 3.291 3.294 3.289 3.293 3.293 3.289 3.293 3.291 3.291 3.291 3.291 3.292 3.289 3.293 3.290 3.289 3.293 3.289 3.290 3.291 3.290 3.290 3.289 3.292 3.288 3.290 3.292 3.287
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 3.290 3.294 3.294 3.293 3.293 3.292 3.295 3.290 3.294 3.295 3.289 3.296 3.292 3.291 3.294 3.292 3.294 3.290 3.294 3.292 3.290 3.295 3.289 3.292 3.294 3.290 3.292 3.291 3.292 3.290 3.291 3.293 3.288 3.293 3.291 3.289 3.293 3.290 3.291 3.290 3.291 3.291 3.288 3.293 3.288 3.289 3.292 3.288 3.291 3.290 3.290 3.289
0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 0.000 3.293 3.291 3.297 3.291 3.293 3.295 3.290 3.294 3.293 3.293 3.292 3.292 3.295 3.289 3.295 3.293 3.289 3.295 3.290 3.292 3.292 3.291 3.292 3.290 3.295 3.290 3.291 3.295 3.288 3.292 3.292 3.290 3.291 3.291 3.292 3.289 3.292 3.292 3.288 3.293 3.290 3.289 3.292 3.289 3.290 3.289 3.291 3.289 3.289 3.293 3.287 3.290
etc
AUTHOR
Utility Programs
Page 74-14
74.7
sdif2ads
DESCRIPTION
Convert files Sound Description Interchange Format (SDIF) to the format usable by Csounds adsyn opcode. As of Csound version 4.10, sdif2ads was available only as a standalone program for Windows console and DOS. Csound U sdif2ads [flags] infilename outfilename -sN apply amplitude scale factor N -pN keep only the first N partials. Limited to 1024 partials. The source partial track indices are used directly to select internal storage. As these can be arbitrary values, the maximum of 1024 partials may not be realized in all cases. -r byte-reverse output file data. The byte-reverse option facilitates transfer across platforms, as Csound's adsyn file format is not portable. If the filename passed to hetro has the extension .sdif, data will be written in SDIF format as 1TRC frames of additive synthesis data. The utility program sdif2ads can be used to convert any SDIF file containing a stream of 1TRC data to the Csound adsyn format. sdif2ads allows the user to limit the number of partials retained, and to apply an amplitude scaling factor. This is often necessary, as the SDIF specification does not, as of the release of sdif2ads, require amplitudes to be within a particular range. sdif2ads reports information about the file to the console, including the frequency range. The main advantages of SDIF over the adsyn format, for Csound users, is that SDIF files are fully portable across platforms (data is big-endian), and do not have the duration limit of 32.76 seconds imposed by the 16 bit adsyn format. This limit is necessarily imposed by sdif2ads. Eventually, SDIF reading will be incorporated directly into adsyn, thus enabling files of any length (subject to system memory limits) to be analysed and processed. Users should remember that the SDIF formats are still under development. While the 1TRC format is now fairly well established, it can still change. For detailed information on the Sound Description Interchange Format, refer to the CNMAT website: http://cnmat.CNMAT.Berkeley.EDU/SDIF Some other SDIF resources (including a viewer) are available via the NC_DREAM website: http://www.bath.ac.uk/~masjpf/NCD/dreamhome.html
AUTHOR
Richard Dobson Somerset, England August, 2000 New in Csound version 4.08
Utility Programs
Page 74-15
Utility Programs
Page 74-16
75
CSCORE
Cscore is a program for generating and manipulating numeric score files. It comprises a number of function subprograms, called into operation by a user-written control program, and can be invoked either as a standalone score preprocessor, or as part of the Csound runtime system: Cscore or scorefilein scorefileout
Csound C [otherflags] orchname scorename The available function programs augment the C language library functions; they can read either standard or pre-sorted score files, can massage and expand the data in various ways, then make it available for performance by a Csound orchestra. The user-written control program is also in C, and is compiled and linked to the function programs (or the entire Csound) by the user. It is not essential to know the C language well to write this program, since the function calls have a simple syntax, and are powerful enough to do most of the complicated work. Additional power can come from C later as the need arises.
Cscore
Page 75.1
75.1
Events, Lists, and Operations

An event in Cscore is equivalent to one statement of a standard numeric score or timewarped score (see any score.srt), stored internally in time-warped format. It is either created in-line, or read in from an existing score file (either format). Its main components are an opcode and an array of pfield values. It is stored somewhere in memory, organized by a structure that starts as follows:
typedef struct { CSHDR h; long op; long pcnt; long strlen; char *strarg; float p2orig; float p3orig; float offtim; float p[1]; } EVENT;
/* /* /* /* /* /*
space-managing header */ opcodet, w, f, I, a, s or e */ number of pfields p1, p2, p3 */ length of optional string argument */ address of optional string argument */ unwarped p2, p3 */
/* storage used during performance */ /* array of pfields p0, p1, p2 */
Any function subprogram that creates, reads, or copies an event will return a pointer to the storage structure holding the event data. The event pointer can be used to access any component of the structure, in the form of e-op or e-p[n]. Each newly stored event will give rise to a new pointer, and a sequence of new events will generate a sequence of distinct pointers that must themselves be stored. Groups of event pointers are stored in an event list, which has its own structure: typedef struct { CSHDR h; int nslots; int nevents; EVENT *e[1]; } EVLIST;
/* max events in this event list */ /* number of events present */ /* array of event pointers e0, e1, e2.. */
Any function that creates or modifies a list will return a pointer to the new list. The list pointer can be used to access any of its component event pointers, in the form of a-e[n]. Event pointers and list pointers are thus primary tools for manipulating the data of a score file. Pointers and lists of pointers can be copied and reordered without modifying the data values they refer to. This means that notes and phrases can be copied and manipulated from a high level of control. Alternatively, the data within an event or group of events can be modified without changing the event or list pointers. The Cscore function subprograms enable scores to be created and manipulated in this way. In the following summary of Cscore function calls, some simple naming conventions are used:
the symbols e, f are pointers to events (notes); the symbols a, b are pointers to lists (arrays) of such events; the letters ev at the end of a function name signify operation on an event; the letter l at the start of a function name signifies operation on a list. the symbol fp is a score input stream file pointer (FILE *); calling syntax description e = createv(n); create a blank event with n pfields int n; e = defev(); defines an event as per the character string e = copyev(f); make a new copy of event f e = getev(); read the next event in the score input file putev(e); write event e to the score output file putstr(); write the string-defined event to score output a = lcreat(n); create an empty event list with n slots int n; a = lappev(a,e); append event e to list a a = lappstrev(a,); append a string-defined event to list a; a = lcopy(b); copy the list b (but not the events) a = lcopyev(b); copy the events of b, making a new list
Cscore
Page 75.2
a = lget(); read all events from score input, up to next s or e a = lgetnext(nbeats); read next nbeats beats from score input float nbeats; a = lgetuntil(beatno); read all events from score input up to beat beatno float beatno; a = lsepf(b); separate the f statements from list b into list a a = lseptwf(b); separate the t,w & f statements from list b into list a a = lcat(a,b); concatenate (append) the list b onto the list a lsort(a); sort the list a into chronological order by p[2] a = lxins(b,); extract notes of instruments (no new events) a = lxtimev(b,from,to); extract notes of time-span, creating new events float from, to; lput(a); write the events of list a to the score output file lplay(a); send events of list a to the Csound orchestra for immediate performance (or print events if no orchestra) relev(e); release the space of event e lrel(a); release the space of list a (but not the events) lrelev(a); release the events of list a, and the list space fp = getcurfp(); get the currently active input scorefile pointer (initially finds the command-line input scorefile pointer) fp = filopen(filename); open another input scorefile (maximum of 5) setcurfp(fp); make fp the currently active scorefile pointer filclose(fp); close the scorefile relating to FILE *fp
Cscore
Page 75.3
75.2
Writing a Main Program

The general format for a control program is:
#include cscore.h cscore() { /* VARIABLE DECLARATIONS /* PROGRAM BODY */ }
*/
The include statement will define the event and list structures for the program. The following C program will read from a standard numeric score, up to (but not including) the first s or e statement, then write that data (unaltered) as output.
#include cscore.h cscore() { EVLIST *a; a = lget(); lput(a); putstr(e); }
/* /* /* /*
a is allowed to point to an event list */ read events in, return the list pointer */ write these events out (unchanged) */ write the string e to output */
After execution of lget(), the variable a points to a list of event addresses, each of which points to a stored event. We have used that same pointer to enable another list function (lput) to access and write out all of the events that were read. If we now define another symbol e to be an event pointer, then the statement
e = a-e[4];
will set it to the contents of the 4th slot in the evlist structure. The contents is a pointer to an event, which is itself comprised of an array of parameter field values. Thus the term ep[5] will mean the value of parameter field 5 of the 4th event in the evlist denoted by a. The program below will multiply the value of that pfield by 2 before writing it out.
#include cscore.h cscore() { EVENT *e; /* a pointer to an event */ EVLIST *a; a = lget(); /* read a score as a list of events */ e = a-e[4]; /* point to event 4 in event list a */ e-p[5] *= 2; /* find pfield 5, multiply its value by 2 */ lput(a); /* write out the list of events */ putstr(e); /* add a score end statement */ } Now consider the following score, in which p[5] contains frequency in Hz. f f I I I e 1 2 1 1 1 0 0 1 4 7 257 257 3 0 3 0 3 0 10 1 7 0 300 1 212 .8 440 10000 256 10000 880 10000
If this score were given to the preceding main program, the resulting output would look like this:
f f I I I e 1 2 1 1 1 0 0 1 4 7 257 257 3 0 3 0 3 0 10 1 7 0 300 1 212 .8 440 10000 512 10000 ; p[5] has become 512 instead of 256. 880 10000
Cscore
Page 75.4
Note that the 4th event is in fact the second note of the score. So far we have not distinguished between notes and function table setup in a numeric score. Both can be classed as events. Also note that our 4th event has been stored in e[4] of the structure. For compatibility with Csound pfield notation, we will ignore p[0] and e[0] of the event and list structures, storing p1 in p[1], event 1 in e[1], etc. The Cscore functions all adopt this convention. As an extension to the above, we could decide to use a and e to examine each of the events in the list. Note that e has not preserved the numeral 4, but the contents of that slot. To inspect p5 of the previous listed event we need only redefine e with the assignment
e = a-e[3];
More generally, if we declare a new variable f to be a pointer to a pointer to an event, the statement
f = &a-e[4];
will set f to the address of the fourth event in the event list a, and *f will signify the contents of the slot, namely the event pointer itself. The expression
(*f)-p[5],
like e-p[5], signifies the fifth pfield of the selected event. However, we can advance to the next slot in the evlist by advancing the pointer f. In C this is denoted by f++. In the following program we will use the same input score. This time we will separate the ftable statements from the note statements. We will next write the three note-events stored in the list a, then create a second score section consisting of the original pitch set and a transposed version of itself. This will bring about an octave doubling. By pointing the variable f to the first note-event and incrementing f inside a while block which iterates n times (the number of events in the list), one statement can be made to act upon the same pfield of each successive event.
#include cscore.h cscore() { EVENT *e,**f; EVLIST *a,*b; int n; a = lget(); b = lsepf(a); lput(b); lrelev(b); e = defev(t 0 120); putev(e); lput(a); putstr(s); putev(e); b = lcopyev(a); n = b-nevents; f = &a-e[1]; while (n--) (*f++)-p[5] *= .5; a = lcat(b,a); lput(a); putstr(e); }
/* declarations. see pp.8-9 in the */ /* C language programming manual */ /* read score into event list a */ /* separate f statements */ /* write f statements out to score */ /* and release the spaces used */ /* define event for tempo statement */ /* write tempo statement to score */ /* write the notes */ /* section end */ /* write tempo statement again */ /* make a copy of the notes in a */ /* and get the number present */ /* iterate the following line n times: */ /* transpose pitch down one octave */ /* now add these notes to original pitches */
Cscore
Page 75.5
The output of this program is:

f f t I I I s t I I I I I I e 1 2 0 1 1 1 0 1 1 1 1 1 1 0 257 0 257 120 1 3 0 4 3 0 7 3 0 120 1 3 4 3 7 3 1 3 4 3 7 3 0 0 0 0 0 0 10 1 7 0 300 1 212 .8 440 10000 256 10000 880 10000 440 256 880 220 128 440 10000 10000 10000 10000 10000 10000
Next we extend the above program by using the while statement to look at p[5] and p[6]. In the original score p[6] denotes amplitude. To create a diminuendo in the added lower octave, which is independent from the original set of notes, a variable called dim will be used.
#include cscore.h cscore() { EVENT *e,**f; EVLIST *a,*b; int n, dim; a = lget(); b = lsepf(a); lput(b); lrelev(b); e = defev(t 0 120); putev(e); lput(a); putstr(s); putev(e); b = lcopyev(a); n = b-nevents; dim = 0; f = &a-e[1]; while (n--){ (*f)-p[6] -= dim; (*f++)-p[5] *= .5; dim += 2000; } a = lcat(b,a); lput(a); putstr(e); }
/* declare two integer variables
*/
/* write out another tempo statement */ /* initialize dim to 0 */ /* subtract current value of dim */ /* transpose, move f to next event */ /* increase dim for each note */
The increment of f in the above programs has depended on certain precedence rules of C. Although this keeps the code tight, the practice can be dangerous for beginners. Incrementing may alternately be written as a separate statement to make it more clear.
while (n--){ (*f)-p[6] -= dim; (*f)-p[5] *= .5; dim += 2000; f++; }
Cscore
Page 75.6
Using the same input score again, the output from this program is:
f f t I I I s t I I I I I I e 1 2 0 1 1 1 0 1 1 1 1 1 1 0 257 0 257 120 1 3 0 4 3 0 7 3 0 120 1 3 4 3 7 3 1 3 4 3 7 3 0 0 0 0 0 0 10 1 7 0 300 1 212 .8 440 10000 256 10000 880 10000 440 256 880 220 128 440 10000 10000 10000 10000 8000 6000 ; Three original notes at ; beats 1,4 and 7 with no dim. ; three notes transposed down one octave ; also at beats 1,4 and 7 with dim.
In the following program the same three-note sequence will be repeated at various time intervals. The starting time of each group is determined by the values of the array cue. This time the dim will occur for each group of notes rather than each note. Note the position of the statement which increments the variable dim outside the inner while block.
#include cscore.h int cue[3]={0,10,17}; /* declare an array of 3 integers */ cscore() { EVENT *e, **f; EVLIST *a, *b; int n, dim, cuecount, holdn; /* declare new variables */ a = lget(); b = lsepf(a); lput(b); lrelev(b); e = defev(t 0 120); putev(e); n = a-nevents; holdn = n; /* hold the value of n to reset below */ cuecount = 0; /* initialize cuecount to 0 */ dim = 0; while (cuecount <= 2) { /* count 3 iterations of inner while */ f = &a-e[1]; /* reset pointer to first event of list a */ n = holdn; /* reset value of n to original note count */ while (n--) { (*f)-p[6] -= dim; (*f)-p[2] += cue[cuecount]; /* add values of cue */ f++; } printf(; diagnostic: cue = %d\n, cue[cuecount]); cuecount++; dim += 2000; lput(a); } putstr(e); }
Here the inner while block looks at the events of list a (the notes) and the outer while block looks at each repetition of the events of list a (the pitch group repetitions). This program also demonstrates a useful trouble-shooting device with the printf function. The semi-colon is first in the character string to produce a comment statement in the resulting score file. In this case the value of cue is being printed in the output to insure that the program is taking the proper array member at the proper time. When output data is wrong or error messages are encountered, the printf function can help to pinpoint the problem.
Cscore
Page 75.7
Using the identical input file, the C program above will generate:
f 1 0 257 10 1 f 2 0 257 7 0 300 1 212 .8 t 0 120 ; diagnostic: cue = 0 I 1 1 3 0 440 10000 I 1 4 3 0 256 10000 I 1 7 3 0 880 10000 ; diagnostic: cue = 10 I 1 11 3 0 440 8000 I 1 14 3 0 256 8000 I 1 17 3 0 880 8000 ; diagnostic: cue = 17 I 1 28 3 0 440 4000 I 1 31 3 0 256 4000 I 1 34 3 0 880 4000 e;
Cscore
Page 75.8
75.3
More Advanced Examples

The following program demonstrates reading from two different input files. The idea is to switch between two 2-section scores, and write out the interleaved sections to a single output file.
./.htmlinclude cscore.h /* CSCORE_SWITCH.C */ cscore() /* callable from either Csound or standalone cscore */ { EVLIST *a, *b; FILE *fp1, *fp2; /* declare two scorefile stream pointers */ fp1 = getcurfp(); /* this is the command-line score */ fp2 = filopen(score2.srt); /* this is an additional score file */ a = lget(); /* read section from score 1 */ lput(a); /* write it out as is */ putstr(s); setcurfp(fp2); b = lget(); /* read section from score 2 */ lput(b); /* write it out as is */ putstr(s); lrelev(a); /* optional to reclaim space */ lrelev(b); setcurfp(fp1); a = lget(); /* read next section from score 1 */ lput(a); /* write it out */ putstr(s); setcurfp(fp2); b = lget(); /* read next sect from score 2 */ lput(b); / * write it out */ putstr(e); }
Finally, we show how to take a literal, uninterpreted score file and imbue it with some expressive timing changes. The theory of composer-related metric pulses has been investigated at length by Manfred Clines, and the following is in the spirit of his work. The strategy here is to first create an array of new onset times for every possible sixteenthnote onset, then to index into it so as to adjust the start and duration of each note of the input score to the interpreted time-points. This also shows how a Csound orchestra can be invoked repeatedly from a run-time score generator.
./.htmlinclude cscore.h /* CSCORE_PULSE.C */
/* program to apply interpretive durational pulse to */ /* an existing score in 3/4 time, first beats on 0, 3, 6 */ static float four[4] = { 1.05, 0.97, 1.03, 0.95 }; static float three[3] = { 1.03, 1.05, .92 }; /* pulse width for 4s*/ /* pulse width for 3s*/
cscore() /* callable from either Csound or standalone cscore */ { EVLIST *a, *b; register EVENT *e, **ep; float pulse16[4*4*4*4*3*4]; /* 16th-note array, 3/4 time, 256 measures */ float acc16, acc1,inc1, acc3,inc3, acc12,inc12, acc48,inc48, acc192,inc192; register float *p = pulse16; register int n16, n1, n3, n12, n48, n192; /* fill the array with interpreted ontimes */ for (acc192=0.,n192=0; n192<4; acc192+=192.*inc192,n192++) for (acc48=acc192,inc192=four[n192],n48=0; n48<4; acc48+=48.*inc48,n48++) for (acc12=acc48,inc48=inc192*four[n48],n12=0;n12<4; acc12+=12.*inc12,n12++) for (acc3=acc12,inc12=inc48*four[n12],n3=0; n3<4; acc3+=3.*inc3,n3++) for (acc1=acc3,inc3=inc12*four[n3],n1=0; n1<3; acc1+=inc1,n1++) for (acc16=acc1,inc1=inc3*three[n1],n16=0; n16<4; acc16+=.25*inc1*four[n16],n16++) *p++ = acc16;
Cscore
Page 75.9
/* for (p = pulse16, n1 = 48; n1--; p += 4)

/* /*
printf(%g %g %g %g %g %g %g %g\n, *p, *(p+1), *(p+2), *(p+3), *(p+1)-*p, *(p+2)-*(p+1), *(p+3)-*(p+2), *(p+4)-*(p+3)); */
/* show vals & diffs */
a = lget(); /* read sect from tempo-warped score */ b = lseptwf(a); /* separate warp & fn statements */ lplay(b); /* and send these to performance */ a = lappstrev(a, s); /* append a sect statement to note list */ lplay(a); /* play the note-list without interpretation */ for (ep = &a-e[1], n1 = a-nevents; n1--; ) { /* now pulse-modify it */ e = *ep++; if (e-op == I) { e-p[2] = pulse16[(int)(4. * e-p2orig)]; e-p[3] = pulse16[(int)(4. * (e-p2orig + e-p3orig))] e-p[2]; } } lplay(a); } /* now play modified list */
As stated above, the input files to Cscore may be in original or time-warped and pre-sorted form; this modality will be preserved (section by section) in reading, processing and writing scores. Standalone processing will most often use unwarped sources and create unwarped new files. When running from within Csound the input score will arrive already warped and sorted, and can thus be sent directly (normally section by section) to the orchestra. A list of events can be conveyed to a Csound orchestra using lplay. There may be any number of lplay calls in a Cscore program. Each list so conveyed can be either timewarped or not, but each list must be in strict p2-chronological order (either from presorting or using lsort). If there is no lplay in a cscore module run from within Csound, all events written out (via putev, putstr or lput) constitute a new score, which will be sent initially to scsort then to the Csound orchestra for performance. These can be examined in the files cscore.out and cscore.srt. A standalone Cscore program will normally use the put commands to write into its output file. If a standalone Cscore program contains lplay, the events thus intended for performance will instead be printed on the console. A note list sent by lplay for performance should be temporally distinct from subsequent note lists. No note-end should extend past the next lists start time, since lplay will complete each list before starting the next (i.e. like a Section marker that doesnt reset local time to zero). This is important when using lgetnext() or lgetuntil() to fetch and process score segments prior to performance.
Cscore
Page 75.10
75.4
Compiling a Cscore Program

A Cscore program can be invoked either as a Standalone program or as part of Csound: or
cscore U pvanal scorename outfilename csound C [otherflags] orchname scorename
To create a standalone program, write a cscore.c program as shown above and test compile it with cc cscore.c. If the compiler cannot find cscore.h, try using I/usr/local/include, or just copy the cscore.h module from the Csound source directory into your own. There will still be unresolved references, so you must now link your program with certain Csound I/O modules. If your Csound installation has created a libcscore.a, you can type
cc o cscore.c lcscore
Else set an environment variable to a Csound directory containing the already compiled modules, and invoke them explicitly:
setenv CSOUND /ti/u/bv/Csound cc o cscore cscore.c $CSOUND/cscoremain.o $CSOUND/cscorefns.o \ $CSOUND/rdscore.o $CSOUND/memalloc.o
The resulting executable can be applied to an input scorefilein by typing:

cscore scorefilein scorefileout
To operate from Csound, first proceed as above then link your program to a complete set of Csound modules. If your Csound installation has created a libcsound.a, you can do this by typing
cc o mycsound cscore.o lcsounc lX11 lm (X11 if your installation included it)
Else copy *.c, *.h and Makefile from the Csound source directory, replace cscore.c by your own, then run make Csound. The resulting executable is your own special Csound, usable as above. The C flag will invoke your Cscore program after the input score is sorted into score.srt. With no lplay, the subsequent stages of processing can be seen in the files cscore.out and cscore.srt.
Cscore
Page 75.11
Cscore
Page 75.12
76
ADDING CSOUND
YOUR OWN
CMODULES
TO
If the existing Csound generators do not suit your needs, you can write your own modules in C and add them to the run-time system. When you invoke Csound on an orchestra and score file, the orchestra is first read by a table-driven translator otran and the instrument blocks converted to coded templates ready for loading into memory by oload on request by the score reader. To use your own C-modules within a standard orchestra you need only add an entry in otrans table and relink Csound with your own code. The translator, loader, and run-time monitor will treat your module just like any other provided you follow some conventions. You need a structure defining the inputs, outputs and workspace, plus some initialization code and some perf-time code. Lets put an example of these in two new files, newgen.h and newgen.c:
typedef struct { /* newgen.h - define a structure */ OPDS h; /* required header */ float *result, *istrt, *incr, *itime, *icontin; /* addr outarg, inargs */ float curval, vincr; /* private dataspace */ long countdown; /* ditto */ } RMP; #include cs.h #include newgen.h /* newgen.c init and perf code */
void rampset(RMP *p) /* at note initialization: */ { if (*p-icontin == 0.) p-curval = *p-istrt; /* optionally get new start value */ p-vincr = *p-incr / esr; /* set s-rate increment per sec. */ p-countdown = *p-itime * esr; /* counter for itime seconds */ } void ramp(RMP *p) /* during note performance: */ { float *rsltp = p-result; /* init an output array pointer */ int nn = ksmps; /* array size from orchestra */ do { *rsltp++ = p-curval; /* copy current value to output */ if (--p-countdown = 0) /* for the first itime seconds, */ p-curval += p-vincr; /* ramp the value */ } while (--nn); }
Now we add this module to the translator table entry.c, under the opcode name rampt:
#include newgen.h void rampset(), ramp(); /* */ opcode { rampt, dspace S(RMP), thread 5, outarg a, inargs iiio, isub rampset, ksub NULL, asub ramp },
Adding your own Cmodules to Csound
Page 76-1
Finally we relink Csound to include the new module. If your Csound installation has created a libcsound.a, you can do this by typing
cc -o mycsound newgen.c entry.c (X11 if included at installation) -lcsound lX11 lm
Else copy *.c, *.h and Makefile from the Csound sources, add newgen.o to the Makefile list OBJS, add newgen.h as a dependency for entry.o, and a new dependency newgen.o: newgen.h, then run make Csound. If your host is a Macintosh, simply add newgen.h and newgen.c to one of the segments in the Csound Project, and invoke the C compiler. The above actions have added a new generator to the Csound language. It is an audio-rate linear ramp function which modifies an input value at a user-defined slope for some period. A ramp can optionally continue from the previous notes last value. The Csound manual entry would look like:
ar rampt istart, islope, itime [, icontin]
istart beginning value of an audio-rate linear ramp. Optionally overridden by a continue flag. islope slope of ramp, expressed as the y-interval change per second. itime ramp time in seconds, after which the value is held for the remainder of the note. icontin (optional) continue flag. If zero, ramping will proceed from input istart . If nonzero, ramping will proceed from the last value of the previous note. The default value is zero. The file newgen.h includes a one-line list of output and input parameters. These are the ports through which the new generator will communicate with the other generators in an instrument. Communication is by address, not value, and this is a list of pointers to floats. There are no restrictions on names, but the input-output argument types are further defined by character strings in entry.c (inargs, outargs). Inarg types are commonly x, a, k, and I, in the normal Csound manual conventions; also available are o (optional, defaulting to 0), p (optional, defaulting to 1). Outarg types include a, k, I and s (asig or ksig). It is important that all listed argument names be assigned a corresponding argument type in entry.c. Also, I-type args are valid only at initialization time, and other-type args are available only at perf time. Subsequent lines in the RMP structure declare the work space needed to keep the code re-entrant. These enable the module to be used multiple times in multiple instrument copies while preserving all data. The file newgen.c contains two subroutines, each called with a pointer to the uniquely allocated RMP structure and its data. The subroutines can be of three types: note initialization, k-rate signal generation, a-rate signal generation. A module normally requires two of these initialization, and either k-rate or a-rate subroutines which become inserted in various threaded lists of runnable tasks when an instrument is activated. The thread-types appear in entry.c in two forms: isub, ksub and asub names; and a threading index which is the sum of isub=1, ksub=2, asub=4. The code itself may reference global variables defined in cs.h and oload.c, the most useful of which are:
extern OPARMS O user-defined user-defined user-defined user-defined user-defined command-line command-line constants sstrcod ; float esr sampling rate float control rate float ksmps int ksmps int nchnls int v flag int m level float float special ekr ensmps ksmps nchnls O.odebug O.msglevel pi, twopi obvious tpidsr twopi / esr float code for string arguments
Page 76-2
FUNCTION
TABLES To access stored function tables, special help is available. The newly defined structure should include a pointer
FUNC *ftp;
initialized by the statement

ftp = ftpfind(p-ifuncno);
where float *ifuncno is an I-type input argument containing the ftable number. The stored table is then at ftp-ftable, and other data such as length, phase masks, cps-to-incr converters, are also accessed from this pointer. See the FUNC structure in cs.h, the ftfind() code in fgens.c, and the code for oscset() and koscil() in opcodes2.c.
A DD I T I O N A L
S PA C E Sometimes the space requirement of a module is too large to be part of a structure (upper limit 65535 bytes), or it is dependent on an I-arg value which is not known until initialization. Additional space can be dynamically allocated and properly managed by including the line
AUXCH auxch;
in the defined structure (*p), then using the following style of code in the init module:
if (p-auxch.auxp == NULL) auxalloc(npoints * sizeof(float), &p-auxch);
The address of this auxiliary space is kept in a chain of such spaces belonging to this instrument, and is automatically managed while the instrument is being duplicated or garbage-collected during performance. The assignment
char *auxp = p-auxch.auxp;
will find the allocated space for init-time and perf-time use. See the LINSEG structure in opcodes1.h and the code for lsgset() and klnseg() in opcodes1.c.
FILE
SHARING
When accessing an external file often, or doing it from multiple places, it is often efficient to read the entire file into memory. This is accomplished by including the line
MEMFIL *mfp;
in the defined structure (*p), then using the following style of code in the init module:
if (p-mfp == NULL) p-mfp = ldmemfile(filname);
where char *filname is a string name of the file requested. The data read will be found between
(char *) p-mfp-beginp; and (char *) p-mfp-endp;
Loaded files do not belong to a particular instrument, but are automatically shared for multiple access. See the ADSYN structure in opcodes3.h and the code for adset() and adsyn() in opcodes3.c.
Page 76-3
STRING
ARGUMENTS
To permit a quoted string input argument (float *ifilnam, say) in our defined structure (*p), assign it the argtype S in entry.c, include another member char *strarg in the structure, insert a line
TSTRARG( rampt, RMP) \
in the file oload.h, and include the following code in the init module:
if (*p-ifilnam == sstrcod) strcpy(filename, unquote(p-strarg));
See the code for adset() in opcodes3.c, lprdset() in opcodes5.c, and pvset() in opcodes8.c.
Page 76-4
77
77.1
A P P E N D I X A: M I S C E L L A N E O U S INFORMATION
Pitch Conversion
Note C-1 C#-1 D-1 D#-1 E-1 F-1 F#-1 G-1 G#-1 A-1 A#-1 B-1 C0 C#0 D0 D#0 E0 F0 F#0 G0 G#0 A0 A#0 B0 C1 C#1 D1 D#1 E1 F1 F#1 G1 G#1 A1 A#1 B1 C2 C#2 D2 D#2 E2 F2 F#2 G2 G#2 A2 A#2 B2 Hz 8.176 8.662 9.177 9.723 10.301 10.913 11.562 12.250 12.978 13.750 14.568 15.434 16.352 17.324 18.354 19.445 20.602 21.827 23.125 24.500 25.957 27.500 29.135 30.868 32.703 34.648 36.708 38.891 41.203 43.654 46.249 48.999 51.913 55.000 58.270 61.735 65.406 69.296 73.416 77.782 82.407 87.307 92.499 97.999 103.826 110.000 116.541 123.471 cpspch 3.00 3.01 3.02 3.03 3.04 3.05 3.06 3.07 3.08 3.09 3.10 3.11 4.00 4.01 4.02 4.03 4.04 4.05 4.06 4.07 4.08 4.09 4.10 4.11 5.00 5.01 5.02 5.03 5.04 5.05 5.06 5.07 5.08 5.09 5.10 5.11 6.00 6.01 6.02 6.03 6.04 6.05 6.06 6.07 6.08 6.09 6.10 6.11 MIDI 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 Note E4 F4 F#4 G4 G#4 A4 A#4 B4 C5 C#5 D5 D#5 E5 F5 F#5 G5 G#5 A5 A#5 B5 C6 C#6 D6 D#6 E6 F6 F#6 G6 G#6 A6 A#6 B6 C7 C#7 D7 D#7 E7 F7 F#7 G7 G#7 A7 A#7 B7 C8 C#8 D8 D#8 Hz 329.628 349.228 369.994 391.995 415.305 440.000 466.164 493.883 523.251 554.365 587.330 622.254 659.255 698.456 739.989 783.991 830.609 880.000 932.328 987.767 1046.502 1108.731 1174.659 1244.508 1318.510 1396.913 1479.978 1567.982 1661.219 1760.000 1864.655 1975.533 2093.005 2217.461 2349.318 2489.016 2637.020 2793.826 2959.955 3135.963 3322.438 3520.000 3729.310 3951.066 4186.009 4434.922 4698.636 4978.032 cpspch 8.04 8.05 8.06 8.07 8.08 8.09 8.10 8.11 9.00 9.01 9.02 9.03 9.04 9.05 9.06 9.07 9.08 9.09 9.10 9.11 10.00 10.01 10.02 10.03 10.04 10.05 10.06 10.07 10.08 10.09 10.10 10.11 11.00 11.01 11.02 11.03 11.04 11.05 11.06 11.07 11.08 11.09 11.10 11.11 12.00 12.01 12.02 12.03 MIDI 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111
Appendix A: Miscellaneous Information
Page 771
Note C3 C#3 D3 D#3 E3 F3 F#3 G3 G#3 A3 A#3 B3 C4 C#4 D4 D#4
Hz 130.813 138.591 146.832 155.563 164.814 174.614 184.997 195.998 207.652 220.000 233.082 246.942 261.626 277.183 293.665 311.127
cpspch 7.00 7.01 7.02 7.03 7.04 7.05 7.06 7.07 7.08 7.09 7.10 7.11 8.00 8.01 8.02 8.03
MIDI 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63
Note E8 F8 F#8 G8 G#8 A8 A#8 B8 C9 C#9 D9 D#9 E9 F9 F#9 G9
Hz 5274.041 5587.652 5919.911 6271.927 6644.875 7040.000 7458.620 7902.133 8372.018 8869.844 9397.273 9956.063 10548.08 11175.30 11839.82 12543.85
cpspch 12.04 12.05 12.06 12.07 12.08 12.09 12.10 12.11 13.00 13.01 13.02 13.03 13.04 13.05 13.06 13.07
MIDI 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127
Page 772
77.2
tone)
Sound Intensity Values (for a 1000 Hz
Dynamics pain fff f p ppp threshold
Intensity (W/m^2) 1 10^-2 10^-4 10^-6 10^-8 10^-12
Level (dB) 120 100 80 60 40 0
Page 773
77.3
soprano a freq (Hz) amp (dB) bw (Hz) soprano e freq (Hz) amp (dB) bw (Hz) soprano I freq (Hz) amp (dB) bw (Hz) soprano o freq (Hz) amp (dB) bw (Hz) soprano u freq (Hz) amp (dB) bw (Hz) alto a freq (Hz) amp (dB) bw (Hz) alto e freq (Hz) amp (dB) bw (Hz) alto I freq (Hz) amp (dB) bw (Hz) alto o freq (Hz) amp (dB) bw (Hz) alto u freq (Hz) amp (dB) bw (Hz)
Formant Values
f1 800 0 80 350 0 60 270 0 60 450 0 70 325 0 50 800 0 80 400 0 60 350 0 50 450 0 70 325 0 50 f2 1150 -6 90 2000 -20 100 2140 -12 90 800 -11 80 700 -16 60 1150 -4 90 1600 -24 80 1700 -20 100 800 -9 80 700 -12 60 f3 2900 -32 120 2800 -15 120 2950 -26 100 2830 -22 100 2700 -35 170 2800 -20 120 2700 -30 120 2700 -30 120 2830 -16 100 2530 -30 170 f4 3900 -20 130 3600 -40 150 3900 -26 120 3800 -22 130 3800 -40 180 3500 -36 130 3300 -35 150 3700 -36 150 3500 -28 130 3500 -40 180 f5 4950 -50 140 4950 -56 200 4950 -44 120 4950 -50 135 4950 -60 200 4950 -60 140 4950 -60 200 4950 -60 200 4950 -55 135 4950 -64 200 tenor a freq (Hz) amp (dB) bw (Hz) tenor e freq (Hz) amp (dB) bw (Hz) tenor I freq (Hz) amp (dB) bw (Hz) tenor o freq (Hz) amp (dB) bw (Hz) tenor u freq (Hz) amp (dB) bw (Hz) bass a freq (Hz) amp (dB) bw (Hz) bass e freq (Hz) amp (dB) bw (Hz) bass I freq (Hz) amp (dB) bw (Hz) bass o freq (Hz) amp (dB) bw (Hz) bass u freq (Hz) amp (dB) bw (Hz) f1 650 0 80 400 0 70 290 0 40 400 0 40 350 0 40 600 0 60 400 0 40 250 0 60 400 0 40 350 0 40 f2 1080 -6 90 1700 -14 80 1870 -15 90 800 -10 80 600 -20 60 1040 -7 70 1620 -12 80 1750 -30 90 750 -11 80 600 -20 80 f3 2650 -7 120 2600 -12 100 2800 -18 100 2600 -12 100 2700 -17 100 2250 -9 110 2400 -9 100 2600 -16 100 2400 -21 100 2400 -32 100 f4 2900 -8 130 3200 -14 120 3250 -20 120 2800 -12 120 2900 -14 120 2450 -9 120 2800 -12 120 3050 -22 120 2600 -20 120 2675 -28 120 f5 3250 -22 140 3580 -20 120 3540 -30 120 3000 -26 120 3300 -26 120 2750 -20 130 3100 -18 120 3340 -28 120 2900 -40 120 2950 -36 120
f1 countertenor freq (Hz) amp (dB) bw (Hz) countertenor freq (Hz) amp (dB) bw (Hz) countertenor freq (Hz) amp (dB) bw (Hz) countertenor freq (Hz) amp (dB) bw (Hz) countertenor freq (Hz) amp (dB) bw (Hz) a 660 0 80 e 440 0 70 I 270 0 40 o 430 0 40 u 370 0 40
f2 1120 -6 90 1800 -14 80 1850 -24 90 820 -10 80 630 -20 60
f3 2750 -23 120 2700 -18 100 2900 -24 100 2700 -26 100 2750 -23 100
f4 3000 -24 130 3000 -20 120 3350 -36 120 3000 -22 120 3000 -30 120
f5 3350 -38 140 3300 -20 120 3590 -36 120 3300 -34 120 3400 -34 120
Page 774
77.4
Window Functions
Windowing functions are used for analysis, and as waveform envelopes, particularly in granular synthesis. Window functions are built in to some opcodes, but others require a function table to generate the window. GEN20 is used for this purpose. The diagram of each window below, is accompanied by the f statement used to generate the it.
HAMMING
f81 0 8192 20 1 1
HANNING
f82 0 8192 20 2 1
Page 775
BARTLETT
f83 0 8192 20 3 1
B L A C KM A N
f84 0 8192 20 4 1
Page 776
B L A C KM A N -H A R R I S
f85 0 8192 20 5 1
GAUSSIAN
f86 0 8192 20 6 1
Page 777
R E C T A N GL E
Note: Vertical scale is exaggerated in this diagram.
f88 0 8192 -20 8 .1
SYNC
f89 0 4096 -20 9 .75
Page 778
77.5
SoundFont2 File Format

Beginning with Csound Version 4.06, Csound supports SoundFont2 sample file format. SoundFont2 (or SF2) is a widespread standard which allows encoding banks of wavetablebased sounds into a binary file. In order to understand the usage of these opcodes, the user must have some knowledge of the SF2 format, so a brief description of this format follows. The SF2 format is made by generator and modulator objects. All current Csound opcodes regarding SF2 support the generator function only. There are several levels of generators having a hierarchical structure. The most basic kind of generator object is a sample. Samples may or may not be be looped, and are associated with a MIDI note number, called the base-key. When a sample is associated with a range of MIDI note numbers, a range of velocities, a transposition (coarse and fine tuning), a scale tuning, and a level scaling factor, the sample and its associations make up a split. A set of splits, together with a name, make up an instrument. When an instrument is associated with a key range, a velocity range, a level scaling factor, and a transposition, the instrument and its associations make up a layer. A set of layers, together with a name, makes up a preset. Presets are normally the final sound-generating structures ready for the user. They generate sound according to the settings of their lower-level components. Both sample data and structure data is embedded in the same SF2 binary file. A single SF2 file can contain up to a maximum of 128 banks of 128 preset programs, for a total of 16384 presets in one SF2 file. The maximum number of layers, instruments, splits, and samples is not defined, and probably is only limited by the computers memory.
Layer
Instrument
Split
Sample
Preset
Layer
Instrument
Split
Sample
Layer
Instrument
Split
Sample
SoundFont2 File Structure
Page 779
77.6
Print Edition Update Procedure

Updated pages only, are available, in Adobe Acrobat (.pdf) format, separately from the complete manual. There are separate sets of files for single- and double-sided printing. The name of the update file for single-sided printing, will be with the Csound version number to which they correspond, ending with up.pdf. For example, the update file for version 3.52 will be called 3_52up.pdf. The files containing the update pages only, for double-sided printing, will follow the same convention as for single-sided printing, except that a 1 will be appended for the odd numbered pages and a 2 for even numbered pages. Example: 3_52up1.pdf and 3_52up2.pdf for Csound version 3.52. There will be as many sets of updates on the server as space permits, in the event the user misses an update before the next one is released. The version of the manual is stated on the title page, in the footer of each page, and in Section 22.6 (Manual Update History). To update an existing manual, print the update file(s) for either single- or double-sided printing, as required. Insert the new pages, and replace the changed pages, as needed, discarding the old pages that have been replaced.
WHERE
TO
GET
T HE
MANUAL
The manual files are available from browser download from the editors website:
http://www.lakewoodsound.com/csound
or via anonymous ftp:

ftp://ftp.csounds.com/manual
All the files are zipped for easy downloading, but the Acrobat files are not compressed. Also available at this site are an HTML Edition, and ASCII text edition, and a Spanish Edition, also in Acrobat format, translated by Servando Valero.
B U G R E PO R T S
We have worked to make these manuals as accurate as possible. Errors, however, will happen. If you find a bug, an error, or omission, please report it to the editor (csound@lakewoodsound.com).
Page 7710
77.7
Manual Update History

Note: Beginning with Version 3.55, page numbers in bold indicate new or added pages. Other page numbers are revised, existing pages.
DATE September, 1998 3 November, 1998
VERSION Version 3.48 Version 3.49
NEW OR CHANGED PAGES All iii, v-xiv, 1:3, 2:1-2:2, 2:13, 5:4, 7:12-7:14, 8:4, 8:168:65, 9:3-9:4, 9:31-9:40, 11:5-11:6, 13:1-13:2, 13:9, 14:4-14:18, 15:18-15:20, 16:4-16:8, 22:6 I, iii, v-xiv, 2:2, 2:16, 8:66, 12:5-12:6, 14:7, 22:6 7:3, 7:14 7:18, 8:59 8:60, 8:66, 9:8 9:9, 9:35, 13:5 13:6 iii, vi xiv, 2:5, 5:3 8:18, 8:20, 22:6 iii, vii-xi, xiii-xiv, 3:1, 4:1, 8:14, 8:15, 8:35, 8:59, 9:32, 9:41, 9:42, 22:6 iii-xvi, 2:9, 5:3, 7:3, 7:19-22, 8:4-6, 8:18, 8:61 8:6770, 9:8-9, 9:21, 9:23 9:36-37, 9:42-44 13:1-2, 13:5-6, 17:1-2, 18:1-2, 22:6 iii, v, vii, xiv, xv, 8:65, 9:14, 17:1, 17:2, 22:6 iii; v-xi; xiii-xv; 5:4; 7:4-7; 7:11; 7:19-22; 8:8-12; 8:49; 8:54; 8:71-72; 9:28; 9:38; 9:42; 12:5; 13:5; 13:6; 13:7; 14:4; 22:6 All I; iii; v-xv; xix; xxiii-xxv; xxviii; xxx; xxxii-xxxiii; 2:2; 5:4-6; 7:6-7; 7:9; 7:12; 7:20; 8:4-5; 8:8; 8:10-11; 8:16; 8:18, 8:28; 8:30; 8:34; 8:34; 8:44-46; 8:65; 8:67-68; 9:1-2; 9:6; 9:10; 9:12; 9:39-40; 9:46; 9:4950; Sec13; 19:1-2; 19:9 I; iii; v-xvi; xix; xxxi; xxxv-xxxvi; 1:5; 2:1-3; 2:10-11; 4:1; 5:4-6; 5:7-8; 7:18; 7:20; 8:5-15; 8:20; 8:24; 8:32; 8:34; 8:37; 8:39; 8:42; 8:60; 8:64; 8:68; 8:70; 8:72; 9:13; 9:23; 9:26; 9:32; 9:34; 9:36-38; 9:39; 9:41; 9:44; 9:48-49; 9:50-58; 10:6; 13:2; 13:6; 13:8; 13:1112; 13:14-19; 14:1-2; 14:4-6; 14:9-10; 14:14-17; 14:19-21; 15:1-7; 15:10-14; 15:16-20; 15:21-22; 16:28; 17:2; 18:2; 19:4-6; 19:10; 20:1; 20:5-7; 20:10-13; 21:2; 22:6; 22:10-11 i-xxxviii; xxxix-xl; 7:6-9; 7:11-12; 7:14-16; 7:18-21; 8:1; 8:4-5; 8:8-9; 8:11; 8:13-15; 8:22-23; 8:63-74; 8:75-76; 9:59-62; 12:7-8; 13:20; 13:21-22; 14:8; 14:11; 22:11-12; 23:1-24 I; v-xl; xli-xlvi; 2:9; 2:16; 5:2; 5:9-10; 6:1; 7:2; 7:18; 7:20; 8:1-3; 8:6-7; 8:18; 8:20; 8:22-23; 8:25; 8:32; 8:40; 8:60; 8:63; 9:1-62; 9:63-64; 12:1; 12:7-8; 12:910; 13:20; 13:22; 14:7; 15:16-22; 19:4-10; 19:11-16; 20:04; 22:11; 23:1-22; (delete 23:23-24)
8 November, 1998 Included in 3.493 release 24 November, 1998 5 January, 1999 21 January, 1999
Version 3.491 Version 3.492 Version 3.493 Version 3.494 Version 3.50
27 January, 1999 24 February, 1999
Version 3.51 Version 3.52
23 March, 1999 20 May, 1999
21 June, 1999
Version 3.55
22 July, 1999
Version 3.56
9 August, 1999
Version 3.57
Page 7711
Manual Update History (Continued)

18 August, 1999 Version 3.58 I; v-xv; iii; xxxi; xxxii; xxxiii; xxxiv; 1:3; 5:4; 7:1; 9:3; 9:13; 9:16; 9:20; 9:25-26; 9:36; 9:38; 9:40; 9:46-64; 14:11-22; 14:23-24; 15:19; 19:11; 22:10; 22:12 All I; ix-xvii; 42-3; 49-7 49-22; 49-23 49-24; 70-5; 76-12; QR10; QR13 I xvii; 2-3 2-5; 3-1; 3-3; 11-11; 11-12; 42-7; 45-1; 50-3; 50-4; 50-5; 50-6; 52-1; 52-2; 68-15; 76-12; QR2; QR11; QR13 All Ixviii; 29-2; 34-1; 35-3; 39-3; 40-11; 42-1142-12; 43-3; 43-12; 43-1343-16; 56-1; 56-3; 58-3; 58-4; 60-6; 60-1160-13; 70-270-24; 77-13; QR10; QR14; QR16 All
16 November, 1999 23 November, 1999 25 February, 2000
Version 4.0 Version 4.01 Version 4.03
1 August, 2000 15 September, 2000
20 March, 2001
Version 4.10
Page 7712
Csound Quick Reference

sr kr ksmps nchnls = = = = strset pset seed ftgen massign ctrlinit iarg iarg iarg iarg iarg, stringtext con1, con2, ... ival ifn, itime, isize, igen, iarga[, iargb, iargz] ichnl, insnum ichnkm, ictlno1, ival1[, ictlno2, ival2[, ictlno3, ival3[, ival32]]
gir
Orchestra Syntax: Variable Data Types

iname kname aname giname gi gkname gk ganame ga wname (init variable - initialization only) (control signal - performance time, control rate) (audio signal - performance time, audio rate) (global init variable - initialization only) (global control signal - performance time, control rate) (global audio signal - performance time, audio rate) (spectral data performance time, control rate)

instr endin
NN

i/k/ar i/k/ar ir i/k/ar = init tival divz iarg iarg ia, ib, isubst

schedule schedwhen schedkwhen turnon insnum, iwhen, idur[, p4, p5, ] ktrigger, kinsnum, kwhen, kdur[, p4, p5, ] ktrigger, kmintim, kmaxnum, kinsnum, kwhen, kdur[, kp4, kp5, ] insnum[, itime]
Instrument Control: Duration Control

ihold turnoff
Version
4.0

Instrument Control: Realtime Performance Control
ir active cpuprc maxalloc prealloc insnum insnum, ipercent insnum, icount insnum, icount

i/kr i/kr kr kr timek times timeinstk timeinsts

ir clockon clockoff readclock inum inum inum

kpitch, kamp kcps, krms ktemp kr kout k/ar kx, ky ar kr kr kr kr pitch pitchamdf tempest follow trigger peak tempo xyin follow2 setctrl control button checkbox sensekey asig, iupdte, ilo, ihi, idbthresh[, ifrqs, iconf, istrt, iocts, iq, inptls, irolloff, iskip] asig, imincps, imaxcps[, icps[, imedi[, idowns [, iexcps]]]] kin, iprd, imindur, imemdur, ihp, ithresh, ihtim, ixfdbak, istartempo, ifn[, idisprd, itweek] asig, idt ksig, kthreshold, kmode k/asig ktempo, istartempo iprd, ixmin, ixmax, iymin, iymax[, ixinit, iyinit] asig, katt, krel inum, kval, itype inum inum inum

(a (a (a (a (a (a > < >= <= == != b b b b b b ? ? ? ? ? ? v1 v1 v1 v1 v1 v1 : : : : : : v2) v2) v2) v2) v2) v2)
Version
4.0

#define #define $NAME. #undef #include NAME # replacement text # NAME(abc) # replacement text # NAME filename

igoto tigoto kgoto goto if if if timout : label label label label ia ka ia istrt
R R R idur
ib kb ib label
igoto kgoto goto
label label label
label

reinit rigoto rireturn label label
Mathematical Operations: Arithmetic and Logic Operations

+ a a a a a a a a a a && b || b + b - b * b / b ^ b % b (no rate restriction) (no rate restriction) (logical AND; not audio-rate) (logical OR; not audio-rate) (no rate restriction) (no rate restriction) (no rate restriction) (no rate restriction) (b not audio-rate) (no rate restriction)
Version
4.0

int(x) int frac(x) frac i(x) abs(x) abs exp(x) exp log(x) log log10(x) log10 sqrt(x) sqrt powoftwo(x) powoftwo logbtwo(x) logbtwo (init-rate or control-rate (init-rate or control-rate (control-rate args only) (no rate restriction) (no rate restriction) (no rate restriction) (no rate restriction) (no rate restriction) (init-rate or control-rate (init-rate or control-rate args only) args only)
args only) args only)

sin(x) sin cos(x) cos tan(x) tan sininv(x) sininv cosinv(x) cosinv taninv(x) taninv sinh(x) sinh cosh(x) cosh tanh(x) tanh (no (no (no (no (no (no (no (no (no rate rate rate rate rate rate rate rate rate restriction) restriction) restriction) restriction) restriction) restriction) restriction) restriction) restriction)

dbamp(x) dbamp ampdb(x) ampdb dbfsamp(x) dbfsamp ampdbfs(x) ampdbfs (init-rate or control-rate args only) (no rate restriction) (init-rate or control-rate args only) (no rate restriction)

rnd(x) rnd birnd(x) birnd (init- or control-rate only) (init- or control-rate only)
Mathematical Operations: Opcode Equivalents of Functions

ar ar i/k/ar i/k/ar ar ar sum product pow taninv2 mac maca asig1, asig2[,asig3asigN] asig1, asig2[,asig3asigN] i/k/aarg, i/k/pow i/k/ax, i/k/ay asig1, ksig1, asig2, ksig2, asig3, asig1, ksig1, asig2, ksig2, asig3,
Version
4.0

octpch(pch) octpch pchoct(oct) pchoct cpspch(pch) cpspch octcps(cps) octcps cpsoct(oct) cpsoct (init- or control-rate (init- or control-rate (init- or control-rate (init- or control-rate (no rate restriction) args args args args only) only) only) only)
Pitch Convertors: Tuning Opcodes

icps icps cps2pch cpsxpch ipch, iequal ipch, iequal, irepeat, ibase

ival ival icps i/kcps icps ioct i/koct ipch i/kpch iamp kaft i/kbend i/kval notnum veloc cpsmidi cpsmidib cpstmid octmidi octmidib pchmidi pchmidib ampmidi aftouch pchbend midictrl [ilow, ihigh] [irange] ifn [irange] [irange] iscal[, ifn] [imin[, imax]] [imin[, imax]] inum[, imin[, imax]]

initc7 initc14 initc21 midic7 midic14 midic21 ctrl7 ctrl14 ctrl21 chanctrl ichan, ictlno, ivalue ichan, ictlno1, ictlno2, ivalue ichan, ictlno1, ictlno2, ictlno3, ivalue ictlno, i/kmin, i/kmax[, ifn] ictlno1, ictlno2, i/kmin, i/kmax[, ifn] ictlno1, ictlno2, ictlno3, i/kmin, i/kmax[, ifn] ichan, ictlno, i/kmin, i/kmax[, ifn] ichan, ictlno1, ictlno2, i/kmin, i/kmax[, ifn] ichan, ictlno1, ictlno2, ictlno3, i/kmin, i/kmax[, ifn] ichnl, ictlno[, ilow, ihigh]
i/kdest i/kdest i/kdest i/kdest i/kdest i/kdest i/kval
Version
4.0

i/k1, , i/k8 i/k1, , i/k16 i/k1, , i/k32 i/k1, , i/k64 k1, , k8 k1, , k16 k1, , k32 k1, , k64 i/k1, , i/k16 i/k1, , i/k32 slider8 slider16 slider32 slider64 slider8f slider16f slider32f slider64f s16b14 s32b14 ichan, ictlnum1, imin1, imax1, init1, ifn1, , ictlnum8, imin8, imax8, init8, ifn8 ichan, ictlnum1, imin1, imax1, init1, ifn1, , ictlnum16, imin16, imax16, init16, ifn16 ichan, ictlnum1, imin1, imax1, init1, ifn1, , ictlnum32, imin32, imax32, init32, ifn32 ichan, ictlnum1, imin1, imax1, init1, ifn1, , ictlnum64, imin64, imax64, init64, ifn64 ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, , ictlnum8, imin8, imax8, init8, ifn8, icutoff8 ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, , ictlnum16, imin16, imax16, init16, ifn16, icutoff16 ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, , ictlnum32, imin32, imax32, init32, ifn32, icutoff32 ichan, ictlnum1, imin1, imax1, init1, ifn1, icutoff1, , ictlnum64, imin64, imax64, init64, ifn64, icutoff64 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, initvalue1, ifn1, , ictlno_msb16, ictlno_lsb16, imin16, imax16, initvalue16, ifn16 ichan, ictlno_msb1, ictlno_lsb1, imin1, imax1, initvalue1, ifn1, , ictlno_msb32, ictlno_lsb32, imin32, imax32, initvalue32, ifn32

kstatus, kchan, kdata1, kdata2 midiin
midiout
kstatus, kchan, kdata1, kdata2

noteon noteoff noteondur noteondur2 moscil midion midion2 ichn, ichn, ichn, ichn, kchn, kchn, kchn, inum, inum, inum, inum, knum, knum, knum, ivel ivel ivel, ivel, kvel, kvel kvel, idur idur kdur, kpause ktrig
Version
4.0

outic outkc outic14 outkc14 outipb outkpb outiat outkat outipc outkpc outipat outkpat nrpn mdelay ichn, inum, ivalue, imin, imax kchn, knum, kvalue, kmin, kmax ichn, imsb, ilsb, ivalue, imin, imax kchn, kmsb, klsb, kvalue, kmin, kmax ichn, ivalue, imin, imax kchn, kvalue, kmin, kmax ichn, ivalue, imin, imax kchn, kvalue, kmin, kmax ichn, iprog, imin, imax kchn, kprog, kmin, kmax ichn, inotenum, ivalue, imin, imax kchn, knotenum, kvalue, kmin, kmax kchan, kparmnum, kparmvalue kstatus, kchan, kd1, kd2, kdelay
MIDI Support: Realtime Messages

mclock mrtmsg ifreq imsgtype
MIDI Support: MIDI Event Extenders

kflag xtratim release iextradur

k/ar k/ar k/ar k/ar k/ar k/ar ar k/ar k/ar k/ar k/ar k/ar line expon linseg linsegr expseg expsegr expsega adsr madsr xadsr mxadsr transeg ia, idur1, ib ia, idur1, ib ia, idur1, ib[, idur2, ic[]] ia, idur1, ib[, idur2, ic[]], irel, iz ia, idur1, ib[, idur2, ic[]] ia, idur1, ib[, idur2, ic[]], irel, iz ia, idur1, ib[, idur2, ic[]] iatt, idec, islev, irel[, idel] iatt, idec, islev, irel[, idel] iatt, idec, islev, irel[, idel] iatt, idec, islev, irel[, idel] ibeg, idur, itype, ival
Version
4.0

Signal Generators: Table Access
i/k/ar i/k/ar i/k/ar kr kr ar table tablei table3 oscil1 oscil1i osciln i/k/andx, ifn[, ixmode[, ixoff[, iwrap]]] i/k/andx, ifn[, ixmode[, ixoff[, iwrap]]] i/k/andx, ifn[, ixmode[, ixoff[, iwrap]]] idel, kamp, idur, ifn idel, kamp, idur, ifn kamp, ifrq, ifn, itimes

k/ar k/ar phasor phasorbnk k/xcps[, iphs] k/xcps, kindx, icnt [, iphs]

k/ar k/ar k/ar k/ar k/ar k/ar oscil oscili oscil3 poscil poscil3 lfo k/xamp, k/xcps, ifn[, iphs] k/xamp, k/xcps, ifn[, iphs] k/xamp, k/xcps, ifn[, iphs] kamp, kcps, ifn[, iphs] kamp, kcps, ifn[, iphs] kamp, kcps[, itype]

ar ar ar buzz gbuzz vco xamp, xcps, knh, ifn[, iphs] xamp, xcps, knh, klh, kr, ifn[, iphs] kamp, kfqc[, iwave][, ipw][, ifn][, imaxd]

ar ar ar adsyn adsynt hsboscil kamod, kfmod, ksmod, ifilcod kamp, kcps, iwfn, ifreqfn, iampfn, icnt[, iphs] kamp, ktone, kbrite, ibasfreq, iwfn, ioctfn[, ioctcnt [, iphs]]
Version
4.0

ar ar ar ar ar ar ar ar ar foscil foscili fmvoice fmbell fmrhode fmwurlie fmmetal fmb3 fmpercfl xamp, xamp, kamp, ifn2, kamp, ifn3, kcps, kcar, kmod, kndx, ifn[, iphs] kcps, kcar, kmod, kndx, ifn[, iphs] kfreq, kvowel, ktilt, kvibamt, kvibrate, ifn1, ifn3, ifn4, ivibfn kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn4, ivfn
kamp, kfreq, kc1, kc2, kvdepth, kvrate, ifn1, ifn2, ifn3, ifn4, ivfn kamp, ifn3, kamp, ifn3, kamp, ifn3, kamp, ifn3, kfreq, kc1, ifn4, ivfn kfreq, kc1, ifn4, ivfn kfreq, kc1, ifn4, ivfn kfreq, kc1, ifn4, ivfn kc2, kvdepth, kvrate, ifn1, ifn2, kc2, kvdepth, kvrate, ifn1, ifn2, kc2, kvdepth, kvrate, ifn1, ifn2, kc2, kvdepth, kvrate, ifn1, ifn2,

ar[, ar2] ar[, ar2] ar ar loscil loscil3 lposcil lposcil3 xamp, kcps, ifn[, ibeg2, iend2]]] xamp, kcps, ifn[, ibeg2, iend2]]] kamp, kfreqratio, kamp, kfreqratio, ibas[, imod1, ibeg1, iend1[, imod2, ibas[, imod1, ibeg1, iend1[, imod2, kloop, kend, ifn[, iphs] kloop, kend, ifn[, iphs]

ar ar ar ar ar fof fof2 fog grain granule xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur[, iphs[, ifmode]] xamp, xfund, xform, koct, kband, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur, kphs, kgliss xamp, xdens, xtrans, xspd, koct, kband, kris, kdur, kdec, iolaps, ifna, ifnb, itotdur[, iphs[, itmode]] xamp, xpitch, xdens, kampoff, kpitchoff, kgdur, igfn, iwfn, imgdur[, igrnd] xamp, ivoice, iratio, imode, ithd, ifn, ipshift, igskip, igskip_os, ilength, kgap, igap_os, kgsize, igsize_os, iatt, idec[, iseed[, ipitch1[, ipitch2[, ipitch3[, ipitch4[, ifnenv]]]]]]]] xamp, xtimewarp, xresample, ifn1, ibeg, iwsize, irandw, ioverlap, ifn2, itimemode xamp, xtimewarp, xresample, ifn1, ibeg, iwsize, irandw, ioverlap, ifn2, itimemode
ar[, ac] ar1, ar2 [,ac1, ac2]
sndwarp sndwarpst
Version
4.0

ar ar ar ar ar ar ar ar pluck wgpluck repluck wgpluck2 wgbow wgflute wgbrass wgclar kamp, icps, iplk, iplk, kamp, kamp, ifn[, kamp, kamp, ifn[, kcps, icps, ifn, imeth[, iparm1, iparm2] iamp, kpick, iplk, idamp, ifilt, axcite xam, icps, kpick, krefl, axcite xam, icps, kpick, krefl kfreq, kpres, krat, kvibf, kvamp, ifn[, iminfreq] kfreq, kjet, iatt, idetk, kngain, kvibf, kvamp, iminfreq[, kjetrf[, kendrf]]] kfreq, iatt, kvibf, kvamp, ifn[, iminfreq] kfreq, kstiff, iatt, idetk, kngain, kvibf, kvamp, iminfreq]

ar ar ar ar ar ar ar ax, ay, az ax, ay, az moog shaker marimba vibes mandol gogobel voice lorenz planet kamp, kfreq, kfiltq, kfiltrate, kvibf, kvamp, iafn, iwfn, ivfn kamp, kfreq, kbeans, kdamp, ktimes[, idecay] kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn, idec kamp, kfreq, kpluck, kdetune, kgain, ksize, ifn[, iminfreq] kamp, kfreq, ihrd, ipos, imp, kvibf, kvamp, ivibfn kamp, kfreq, kphoneme, kform, kvibf, kvamp, ifn, ivfn ks, kr, kb, kh, ix, iy, iz, iskip kmass1, kmass2, ksep, ix, iy, iz, ivx, ivy, ivz, idelta, ifriction

ar kfr, kap ar ar pvoc pvread pvbufread pvinterp pvcross tableseg tablexseg vpvoc pvadd ktimpnt, kfmod, ifilcod, ifn, ibins[, ibinoffset, ibinincr, iextractmode, ifreqlim, igatefn] ktimpnt, ifile, ibin ktimpnt, ifile ktimpnt, kfmod, ifile, kfreqscale1, kfreqscale2, kampscale1, kampscale2, kfreqinterp, kampinterp ktimpnt, kfmod, ifile, kamp1, kamp2[, ispecwp] ifn1, idur1, ifn2[, idur2, ifn3[]] ifn1, idur1, ifn2[, idur2, ifn3[]] ktimpnt, kfmod, ifile[, ispecwp] ktimpnt, kfmod, ifilcod, ifn, ibins[, ibinoffset, ibinincr, iextractmode, ifreqlim, igatefn]
ar ar
10
Version
4.0

krmsr, krmso, kerr, kcps ar ar lpread ktimpnt, ifilcod[, inpoles[, ifrmrate]]
lpreson lpfreson lpslot lpinterp
asig asig, kfrqratio islot islot1, islot2, kmix

k/ar k/ar k/ar i/k/ar i/k/ar i/k/ar i/k/ar i/k/ar i/k/ar i/k/ar i/k/ar i/k/ar i/k/ar i/k/ar ar ar rand randh randi linrand trirand exprand bexprnd cauchy pcauchy poisson gauss weibull betarand unirand pinkish noise k/xamp[, iseed[, isize]] k/xamp, k/xcps[, iseed[, isize]] k/xamp, k/xcps[, iseed[, isize]] krange krange krange krange kalpha kalpha klambda krange ksigma, ktau krange, kalpha, kbeta krange xin[, imethod, inumbands, iseed, iskip] xamp, kbeta

nsamp(x) nsamp ftlen(x) ftlen ftlptim(x) ftlptim ftsr(x) ftsr tableng (init-rate (init-rate (init-rate (init-rate i/kfn args args args args only) only) only) only)
i/kr
11
Version
4.0

k/ar k/ar tablekt tableikt k/xndx, i/kfn[, ixmode[, ixoff[, iwrap]]] k/xndx, kfn[, ixmode[, ixoff[, iwrap]]]

tablew tablewkt tableiw tableigpw tablegpw tableimix tablemix tableicopy tablecopy tablera tablewa i/k/asig, i/k/andx, ifn[, ixmode[, ixoff[, iwgmode]]] k/asig, k/andx, kfn[, ixmode[, ixoff[, iwgmode]]] isig, indx, ifn[, ixmode[, ixoff[, iwrap]]] ifn kfn idft, idoff, ilen, is1ft, is1off, is1g, is2ft, is2off, is2g kdft, kdoff, klen, ks1ft, ks1off, ks1g, ks2ft, ks2off, ks2g idft, isft kdft, ksft kfn, kstart, koff kfn, asig, koff
ar kstart
12
Version
4.0

kr kr kr ar kr ar kr ar kr ar ar ar ar ar ar ar ar ar ar ar ar ar ar alow, ahigh, aband ar1, ar2 ar ar ar ar k/ar ar ar ar portk port tonek tone atonek atone resonk reson aresonk areson tonex atonex resonx resonr resonz resony lowres lowresx vlowres lowpass2 biquad rezzy moogvcf svfilt hilbert butterhp butterlp butterbp butterbr filter2 zfilter2 lpf18 tbvcf ksig, ksig, ksig, asig, ksig, asig, ksig, asig, ksig, asig, asig, asig, asig, asig, asig, asig, asig, asig, asig, asig, asig, asig, asig, asig, khtim[, isig] ihtim[, isig] khp[, iskip] khp[, iskip] khp[, iskip] khp[, iskip] kcf, kbw[, iscl, iskip] kcf, kbw[, iscl, iskip] kcf, kbw[, iscl, iskip] kcf, kbw[, iscl, iskip] khp[, inumlayer, iskip] khp[, inumlayer, iskip] kcf, kbw[, inumlayer, iscl, iskip] kcf, kbw[,iscl, iskip] kcf, kbw[,iscl, iskip] kbf, kbw, inum, ksep[,iscl, iskip] kcutoff, kresonance[, iskip] kcutoff, kresonance[, inumlayer, iskip] kfco, kres, iord, ksep kcf, kq[, iskip] kb0, kb1, kb2, ka0, ka1, ka2[, iskip] xfco, xres[, imode] xfco, xres[, iscale] kcf, kq[, iscl]
asig asig, kfreq[, iskip] asig, kfreq[, iskip] asig, kfreq, kband[, iskip] asig, kfreq, kband[, iskip] k/asig, iM, iN, ib0, ib1, , ibM, ia1, ia2, , iaN asig, kdamp, kfreq, iM, iN, ib0, ib1, , ibM, ia1, ia2, , iaN asig, kfco, kres, kdist asig, xfco, xres, kdist, kasym

ar ar ar nlfilt pareq dcblock ain, ka, kb, kd, kL, kC asig, kc, iv, iq, imode asig[, ig]
13
Version
4.0

k/ar k/ar k/ar k/ar linen linenr envlpx envlpxr k/xamp, k/xamp, k/xamp, k/xamp, irind]] irise, irise, irise, irise, idur, idec, idur, idur, idec iatdec idec, ifn, iatss, iatdec[, ixmod] idec, ifn, iatss, iatdec[, ixmod[,

kr ar ar ar rms gain balance dam asig[, ihp, iskip] asig, krms[, ihp, iskip] asig, acomp[, ihp, iskip] ain, kthreshold, icomp1, icomp2, rtime, ftime

i/k/ar i/k/ar i/k/ar wrap mirror limit i/k/asig, i/k/klow, i/k/khigh i/k/asig, i/k/klow, i/k/khigh i/k/asig, i/k/klow, i/k/khigh

ar ar ar ar ar ar ar ar ar ar delayr delayw delay delay1 deltap deltapi deltapn deltap3 multitap vdelay vdelay3 idlt[, iskip] asig asig, idlt[, iskip] asig[, iskip] kdlt xdlt xnumsamps xdlt asig, itime1, igain1, itime2, igain2 asig, adel, imaxdel[, iskip] asig, adel, imaxdel[, iskip]

ar ar ar ar ar ar a1, a2 reverb reverb2 nreverb comb alpass nestedap babo asig, krvt[, iskip] asig, ktime, khdif[, iskip] asig, ktime, khdif[, iskip][,inumCombs, ifnCombs][, inumAlpas, ifnAlpas] asig, krvt, ilpt[, iskip][, insmps] asig, krvt, ilpt[, iskip][, insmps] asig, imode, imaxdel, idel1, igain1[, idel2, igain2[, idel3, igain3]] asig, ksrcx, ksrcy, ksrcz, irx, iry, irz[, idiff[, ifno]]
14
Version
4.0

ar ar ar ar wguide1 wguide2 streson nlalp asig, kfreq, kcutoff, kfeedback asig, kfreq1, kfreq2, kcutoff1, kcutoff2, kfeedback1, kfeedback2 asig, kfr, ifdbgain asig, klcf, knlcf [, iskip[, iupdm]]
Signal Modfiers: Special Effects

ar ar ar ar ar harmon flanger distort1 phaser1 phaser2 asig, kestfrq, kmaxvar, kgenfreq1, kgenfreq2, imode, iminfrq, iprd asig, adel, kfeedback[, imaxd] asig[, kpregain[, kpostgain[, kshape1[, kshape2]]]] asig, kfreq, iord, kfeedback[, iskip] asig, kfreq, iord, imode, ksep, kfeedback

ar1[, ar2[, ar3[, ar4]]] ar convolve ain, ifilcod, ichannel
cross2
ain1, ain2, isize, ioverlap, iwin, kbias

a1, a3, a1, a1, a3, a1, a1, a3, a1, a3, a1, a3, a2, a4 a2 a2, a4 a2 a2, a4 a2, a4 a2, a4 pan locsig locsig locsend locsend space spsend asig, ifn, ktime, kreverbsend[, kx, ky] asig, kx, ky, ifn[, imode[, ioffset]] asig, kdegree, kdistance, kreverbsend asig, kdegree, kdistance, kreverbsend
k1 aleft, aright
spdist hrtfer
ifn, ktime[, kx, ky] asig, kaz, kelev, HRTFcompact
15
Version
4.0

kr ar ar k/ar k/ar k/ar i/k/ar ar downsamp upsamp interp integ diff samphold ntrpol fold asig[, iwlen] ksig ksig[, iskip] k/asig[, iskip] k/asig[, iskip] x/asig, k/xgate[, ival, ivstor] i/k/asig1, i/k/asig2, i/k/kpoint[, imin, imax] asig, kincr
Zak Patch System

zakinit ziw zkw zaw ziwm zkwm zawm zir zkr zar zarg zkmod zamod zkcl zacl isizea, isizek isig, indx ksig, kndx asig, kndx isig, indx[, imix] ksig, kndx[, kmix] asig, kndx[, kmix] indx kndx kndx kndx, kgain ksig, kzkmod asig, kzamod kfirst, klast kfirst, klast
ir kr ar ar kr ar
Operations Using Spectral Data-Types

wsig wsig wsig wsig wsig koct, kamp ksum wsig specaddm specdiff specscal spechist specfilt specptrk specsum specdisp spectrum wsig1, wsig2[, imul2] wsigin wsigin, ifscale, ifthresh wsigin wsigin, ifhtim wsig, kvar, ilo, ihi, istrt, idbthresh, inptls, irolloff[, iodd, iconfs, interp, ifprd, iwtflg] wsig[, interp] wsig, iprd[, iwtflg] xsig, iprd, iocts, ifrqa, iq[, ihann, idbout, idsprd, idsinrs]
16
Version
4.0

a1 a1, a2 a1, a2, a3, a4 a1, a2, a3, a4, a5, a6 a1, a2, a3, a4, a5, a6, a7, a8 a1 a1, a2 a1, a2, a3, a4 a1[,a2 [,a3,a4]] in ins inq inh ino
soundin soundin soundin diskin
ifilcod[, iskptim[, iformat]] ifilcod[, iskptim[, iformat]] ifilcod[, iskptim[, iformat]] ifilcod, kpitch[, iskiptim[, iwraparound[, iformat]]]

out outs1 outs2 outs outq1 outq2 outq3 outq4 outq outh outo soundout soundouts asig asig asig asig1, asig asig asig asig asig1, asig1, asig1, asig1, asig1,
asig2
asig2, asig3, asig4 asig2, asig3, asig4, asig5, asig6 asig2, asig3, asig4, asig5, asig6, asig7, asig8 ifilcod[, iformat] asig2, ifilcod[, iformat] (**Not implemented***)
17
Version
4.0

Signal Input and Output: File I/O
dumpk dumpk2 dumpk3 dumpk4 readk readk2 readk3 readk4 fout foutk fouti foutir ihandle fiopen fin fink fini vincr clear ksig, ifilname, iformat, iprd ksig1, ksig2, ifilname, iformat, iprd ksig1, ksig2, ksig3, ifilname, iformat, iprd ksig1, ksig2, ksig3, ksig4, ifilname, iformat, iprd ifilname, iformat[, ipol] ifilname, iformat[, ipol] ifilname, iformat[, ipol] ifilname, iformat[, ipol] ifilename, iformat, aout1[, aout2, aout3,...,aoutN] ifilename, iformat, aout1[, aout2, aout3,...,aoutN] ihandle, iformat, iflag, iout1[, iout2, iout3,....,ioutN] ihandle, iformat, iflag, iout1[, iout2, out3,....,ioutN] ifilename,imode ifilename, iskipframes, iformat, ain1[, ain2, ain3,...,ainN] ifilename, iskipframes, iformat, kin1[, kin2, kin3,...,kinN] ifilename, iskipframes, iformat, in1[, in2, in3,...,inN] asig, aincr avar1[,avar2, avar3,,avarN]
ksig k1, k2 k1,k2,k3 k1,k2, k3,k4

ir ir ir ir filelen filesr filenchnls filepeak ifilcod ifilcod ifilcod ifilcod[, ichnl]

print display dispfft printk printks printk2 iarg[, iarg, ...] xsig, iprd[, inprds[, iwtflg]] xsig, iprd, iwsiz[, iwtyp[, idbouti[, iwtflg]]] kval, ispace[, itime] txtstring, itime, kval1, kval2, kval3, kval4 kvar[, numspaces]
18
Version
4.0

Score Syntax: Statements
f f0 b t a i s m n r e table number action time size GEN routine arg1[ arg2...arg...] action time (Dummy f-table for padding score sections with silence and reporting on progress of long running jobs). base clock time (Effective prior to score sorting. This time base is pre-warped.) 0 initial tempo time in beats tempo2[time in beats tempo3 time in...] 0 begin time advance in beats duration of time advance in beats instrument number start duration [p4 p5 p...] (marks end of section and restarts score counting from time 0) score location name (marks a score section with a name) score location name (named score section is re-read into the score file at this location) integer repeat count a macro name (begins a new repeating sections) (marks end of score - optional)
Score Syntax: P-Field Substitution

. + ^+x ^-x npx ppx < > ) ( ~ (carries same p-field value from preceding i statement with like i instrument #) (determines current start from sum of preceeding durations by adding p2 + p3 from previous i statement. legal in p2 only.) i (determines current start of instrument from sum of preceeding written event by adding last p2 to x. legal in p2 only.) (determines current start of instrument from sum of preceeding written event by subtracting x from last p2. legal in p2 only.) (replace with p-field(x) value from next note statement illegal in p1 p2 p3.) (replace with p-field(x) value from previous note statement illegal in p1 p2 p3.) (p-field replaced by value derived from linear interpolation between previous and subsequent anchor values in same p-field. illegal in p1 p2 p3) (p-field replaced by value derived from linear interpolation between previous and subsequent anchor values in same p-field. illegal in p1 p2 p3) (p-field replaced by value derived from exponential interpolation between previous and subsequent anchor values in same p-field. illegal in p1 p2 p3) (p-field replaced by value derived from exponential interpolation between previous and subsequent anchor values in same p-field. illegal in p1 p2 p3) (p-field replaced by value derived from random value in the range between previous and subsequent anchor values in same p-field. illegal in p1 p2 p3)
19
Version
4.0

Score Syntax: Expressions
[x+y] + [x-y] [x*y] * [x/y] / [x%y] % [x^y] ^ [@ [ x] [@@x] (add value x to value y within a p-field. Note expressions must be in [brackets]) ] (subtract value y from value x within a p-field. Note expressions must be in [brackets]) ] (multiply value x by value y within a p-field. Note expressions must be in [brackets]) ] (divide value x by value y within a p-field. Note expressions must be in [brackets]) ] (value x remainder value y within a p-field. Note expressions must be in [brackets]) ] (power of value x to value y within a p-field. Note expressions must be in [brackets]) ] (next power-of-two greater than or equal to x. Note expressions must be in [brackets]) ] (next power-of-two-plus-one greater than or equal to x. Note expressions must be in [brackets]) ]
Score Syntax: Macros

#define #define $NAME. #undef #include NAME # replacement text # NAME(abc) # replacement text # NAME filename
20
Version
4.0

GEN Routines: Sine/Cosine Generators
f f f f # # # # time time time time size size size size 9 10 19 11 pna str1 pna nh stra str2 stra lh phsa str3 phsa r pnb str4 dcoa strb pnb phsb strb phsb dcob
GEN Routines: Line/Exponential Segment Generators

f f f f f f f # # # # # # # time time time time time time time size size size size size size size 5 6 7 8 16 25 27 a a a a beg x1 x1 n1 n1 n1 n1 dur y1 y1 b b b b type x2 x2 n2 n2 n2 n2 end y2 y2 c c c c x3 x3 n3 n3 ... ... d d
GEN Routines: File Access

f f f # # # time time time size size 0 1 23 28 filcod skiptime filename.txt filcod format channel
GEN Routines: Numeric Value Access

f f # # time time size size 2 17 v1 x1 v2 a v3 x2 b x3 c
GEN Routines: Window Functions

f # time size 20 window max op
GEN Routines: Random Functions

f # time size 21 type lvl arg1 arg2
GEN Routines: Waveshaping

f f f f # # # # time time time time size size size size 3 13 14 15 xval1 xint xint xint xval2 xamp xamp xamp c0 h0 h0 h0 c1 h1 h1 phs0 c2 h2 h2 h1 cn hn hn phs1 h2
phs2
GEN Routines: Amplitude Scaling

f f # # time time size size 4 12 source# xint sourcemode
21
Version
4.0

Command Line Flags: Generic
-I -n -i -o -b -B -A -W -J -h -c -a -8 -u -s -l -f -r -k -v -m i-time only orch run no sound onto disk sound input filename fnam sound output filename fnam sample frames (or -kprds) per software sound I/O buffer samples per hardware sound I/O buffer create an AIFF format output soundfile create a WAV format output soundfile create an IRCAM format output soundfile no header on output soundfile 8-bit signed_char sound samples alaw sound samples 8-bit unsigned_char sound samples ulaw sound samples short_int sound samples long_int sound samples float sound samples orchestra srate override orchestra krate override verbose orch translation tty message level. N = Sum of: 1 = note amps, 2 = out-of-range msg, 4 = warnings suppress all displays suppress graphics, use ASCII displays create Postscript displays of any display score is in Scot format extract from score.srt using extract file fnam use uninterpreted beats of the score, initially at tempo N read Line-oriented real-time score events from device dnam read MIDI real-time events from device dnam read MIDI file event stream from file fnam MIDI sustain pedal threshold (N = 0 - 128) continually rewrite header while writing soundfile (WAV/AIFF) print a heartbeat character at each soundfile write generates a . every time a buffer is written. reports the size in seconds of the output. sounds a bell for every buffer of the output written. notify (ring the bell) when score or MIDI file is done terminate the performance when MIDI file is done defer GEN01 soundfile loads until performance time List opcodes in this version List opcodes and arguments in this version Log all text output to lognam Derive console messages from database fnam Switch off peak chunks.
fnam fnam N N
N N N
-d -g -G -S -x fnam -t N -L dnam -M dnam -F fnam -P N -R -H/H1 -H2 -H3 -H4 -N -T -D -z -z1 -- lognam -j fnam -K
22
Version
4.0

Command Line Flags: Utility Invocation
-U -U -U -U -U -U -C sndinfo hetro lpanal pvanal cvanal pvlook run run run run run run use utility program sndinfo utility program hetro utility program lpanal utility program pvanal utility program cvanal utility program pvlook Cscore processing of scorefile
Command Line Flags: PC and Windows-Specific

-j -J -K -q
num num num num
-p num -O -e -y -E -Q num -Y -*
set the number of console text rows (default 25) set the number of console text columns (default 80) enables MIDI IN. num (optional) = MIDI IN port device id number WAVE OUT device id number (use only if more WAVE devices are installed) number of WAVE OUT buffers (default 4; max. 40) suppresses all console text output for better real-time performance allows any sample rate (to use only with WAVE cards supporting this feature) doesnt wait for keypress on exit allows graphic display for WCSHELL by Riccardo Bianchini enable MIDI OUT. num (optional) = MIDI OUT port device id number suppresses real-time WAVE OUT for better MIDI OUT timing performance yields control to the system until audio output buffer is full
Command Line Flags: Macintosh-Specific

-q -Q -X -V -E -p -e -w -y -Y
sampdir analdir snddir num num num num num
set the directory for finding samples set the directory for finding analyses set the directory for saving sound files set screen buffer size set number of graphs saved play on finishing set rescaling factor set recording of MIDI data set rate for progress display set rate for profile display
23
Version
4.0

Utilities: Analysis File Generation
hetro -sr n -c n -b n -d n -f n -h -M -m -n infilename infilename infilename infilename infilename infilename infilename infilename infilename infilename infilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename Hetrodyne analysis sample rate Hetrodyne analysis channel number Hetrodyne analysis segment begin time Hetrodyne analysis segment duration Hetrodyne analysis beginning frequency Hetrodyne analysis number of partials Hetrodyne analysis maximum amplitude Hetrodyne analysis minimum amplitude Hetrodyne analysis number of breakpoints Hetrodyne analysis use third order low-pass filter with fc of n LPC analysis write filter pole instead of coeffecients LPC analysis sample rate LPC analysis channel number LPC analysis segment begin time LPC analysis segment duration LPC analysis number of poles LPC analysis hop size in samples LPC analysis text string for comments LPC analysis lowest frequency LPC analysis highest frequency LPC analysis verbosity level of terminal messages STFT STFT STFT STFT STFT STFT STFT FFT FFT FFT FFT analysis analysis analysis analysis analysis analysis analysis analysis analysis analysis analysis
n n n n
-l n lpanal -a -s -c -b -d -p -h -C -P -Q -v
n n n n n n s n n n
infilename infilename infilename infilename infilename infilename infilename infilename infilename infilename
pvanal
-s -c -b -d -n -w -h -s -c -b -d
n n n n n n n n n n n
infilename infilename infilename infilename infilename infilename infilename infilename infilename infilename infilename
outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename outfilename
sample rate channel number segment begin time segment duration frame size window overlap factor hop size in samples sample rate channel number segment begin time segment duration
cvanal
24
Version
4.0

Utilities: File Queries
sndinfo pvlook -bb n -eb n -bf n -ef n -i soundfilename infilename infilename infilename infilename infilename get info about one or more sound files soundfilename STFT analysis file formatted text output beginning bin number STFT analysis file formatted text output ending bin number STFT analysis file formatted text output beginning frame number STFT analysis file formatted text output ending frame number STFT analysis file formatted text output as integers
25
Version
4.0

Csound Manual

Încărcat de

Informații document

Descriere originală:

Drepturi de autor

Formate disponibile

Partajați acest document

Partajați sau inserați document

Opțiuni de partajare

Vi se pare util acest document?

Este necorespunzător acest conținut?

Drepturi de autor:

Formate disponibile

Csound Manual

Încărcat de

Drepturi de autor:

Formate disponibile

The Public Csound Reference Manual

CANONICAL VERSION 4.10

The Public Csound Reference Manual Version 4.10

The Public Csound Reference Manual Version 4.10

This page intentionally left blank.

The Public Csound Reference Manual Version 4.10

2 SYNTAX OF THE ORCHESTRA

3 ORCHESTRA SYNTAX: ORCHESTRA HEADER STATEMENTS

4 ORCHESTRA SYNTAX: INSTRUMENT BLOCK STATEMENTS

5 ORCHESTRA SYNTAX: VARIABLE INITIALIZATION

6 INSTRUMENT CONTROL: INSTRUMENT INVOCATION

The Public Csound Reference Manual Version 4.10

7 INSTRUMENT CONTROL: DURATION CONTROL STATEMENTS

8 INSTRUMENT CONTROL: REAL-TIME PERFORMANCE CONTROL

9 INSTRUMENT CONTROL: TIME READING

10 INSTRUMENT CONTROL: CLOCK CONTROL

11 INSTRUMENT CONTROL: SENSING AND CONTROL

12 INSTRUMENT CONTROL: CONDITIONAL VALUES

13 INSTRUMENT CONTROL: MACROS

14 INSTRUMENT CONTROL: PROGRAM FLOW CONTROL

15 INSTRUMENT CONTROL: REINITIALIZATION

16 MATHEMATICAL OPERATIONS: ARITHMETIC AND LOGIC OPERATIONS

17 MATHEMATICAL OPERATIONS: MATHEMATICAL FUNCTIONS

The Public Csound Reference Manual Version 4.10

18 MATHEMATICAL OPERATIONS: TRIGONOMETRIC FUNCTIONS

19 MATHEMATICAL OPERATIONS: AMPLITUDE FUNCTIONS

20 MATHEMATICAL OPERATIONS: RANDOM FUNCTIONS

21 MATHEMATICAL FUNCTIONS: OPCODE EQUIVALENTS OF FUNCTIONS

22 PITCH CONVERTERS: FUNCTIONS

23 PITCH CONVERTERS: TUNING OPCODES

24 MIDI SUPPORT: CONVERTERS

25 MIDI SUPPORT: CONTROLLER INPUT

26 MIDI SUPPORT: SLIDER BANKS

27 MIDI SUPPORT: GENERIC I/O

28 MIDI SUPPORT: NOTE-ON/NOTE-OFF

The Public Csound Reference Manual Version 4.10

29 MIDI SUPPORT: MIDI MESSAGE OUTPUT

30 MIDI SUPPORT: REAL-TIME MESSAGES

31 MIDI SUPPORT: EVENT EXTENDERS

32 SIGNAL GENERATORS: LINEAR AND EXPONENTIAL GENERATORS

33 SIGNAL GENERATORS: TABLE ACCESS

34 SIGNAL GENERATORS: PHASORS

35 SIGNAL GENERATORS: BASIC OSCILLATORS

36 SIGNAL GENERATORS: DYNAMIC SPECTRUM OSCILLATORS

37 SIGNAL GENERATORS: ADDITIVE SYNTHESIS/RESYNTHESIS

38 SIGNAL GENERATORS: FM SYNTHESIS

The Public Csound Reference Manual Version 4.10

39 SIGNAL GENERATORS: SAMPLE PLAYBACK

40 SIGNAL GENERATORS: GRANULAR SYNTHESIS

41 SIGNAL GENERATORS: SCANNED SYNTHESIS

42 SIGNAL GENERATORS: WAVEGUIDE PHYSICAL MODELING

43 SIGNAL GENERATORS: MODELS AND EMULATIONS

44 SIGNAL GENERATORS: STFT RESYNTHESIS (VOCODING)

45 SIGNAL GENERATORS: LPC RESYNTHESIS

The Public Csound Reference Manual Version 4.10

46 SIGNAL GENERATORS: RANDOM (NOISE) GENERATORS

47 FUNCTION TABLE CONTROL: TABLE QUERIES

48 FUNCTION TABLE CONTROL: TABLE SELECTION

49 FUNCTION TABLE CONTROL: READ/WRITE OPERATIONS

50 SIGNAL MODIFIERS: STANDARD FILTERS

51 SIGNAL MODIFIERS: SPECIALIZED FILTERS

52 SIGNAL MODIFIERS: ENVELOPE MODIFIERS

53 SIGNAL MODIFIERS: AMPLITUDE MODIFIERS